From volga629 at networklab.ca Sun Jul 1 00:28:03 2018 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Sun, 01 Jul 2018 01:28:03 -0300 Subject: [OpenSIPS-Users] cluster presence In-Reply-To: <80fe446c-ba05-9947-bac4-56a1f962865b@opensips.org> References: <1528294844.2069.1@smtp.networklab.ca> <81a4efef-f696-2ac0-80fb-de84d12d6862@opensips.org> <1528375427.2069.3@smtp.networklab.ca> <80fe446c-ba05-9947-bac4-56a1f962865b@opensips.org> Message-ID: <1530419283.6843.0@smtp.networklab.ca> Hello Bogdan, I checked database connection and it looks normal ping less then sec to database. I monitored load on database nodes and didn't notcied any extra load on them. Right now I see 2 types of messages on all cluster nodes. Jun 30 22:56:45 aitossbc01 /usr/sbin/opensips[20245]: WARNING:core:timer_ticker: timer task already scheduled 100410 ms ago (now 700510 ms), skipping execution Jun 30 22:56:45 aitossbc01 /usr/sbin/opensips[20245]: WARNING:core:timer_ticker: timer task already scheduled 191700 ms ago (now 700510 ms), delaying execution and Jun 30 23:14:36 aitossbc02 /usr/sbin/opensips[9376]: WARNING:core:utimer_ticker: utimer task already scheduled 3880 ms ago (now 113380 ms), delaying execution Jun 30 23:14:36 aitossbc02 /usr/sbin/opensips[9376]: WARNING:core:utimer_ticker: utimer task already scheduled 5880 ms ago (now 113380 ms), delaying execution Also that not affecting call or call quality. Only filling up logs. volga629 On Thu, Jun 14, 2018 at 6:48 AM, Bogdan-Andrei Iancu wrote: > Hi Volga, > > How large is the presentity data set in your system ? I'm asking as > the routine that seems to be slow queries the presentity table in > order to get the expired presentities - this is done with or without > clustering. Still, in clustering, all the nodes are doing this query, > putting extra stress on the DB. > > Now, for each expire presentity, OpenSIPS has to send a NOTIFY to its > subscribers - and here, having clustering enable, it is a difference. > IF you have 3 nodes, so 3 shared tags, OpenSIPS will do 3 queries > (one per tag) in order to fetch from active_watchers the subscribers > with the tag looking for the presentity. So, who large is the > subscriber's data set ? > > Do you notice any extra DB load when the cleanup timer kicks in ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 06/07/2018 03:43 PM, volga629 at networklab.ca wrote: >> Hello Bogdan-Andrei, >> Yes those messages start showing up when cluster enabled. Standalone >> mode all works no issues. >> Right now in opensips cluster we have 2 active one backup. In PgSQL >> 5 nodes 3 active 2 backup. In Mongodb cluster: 2 mongos 3 config 2 >> shred. >> We use postgres BDR cluster and MongoDB Cluster. >> >> Here are configuration >> >> #### Presence >> loadmodule "presence.so" >> loadmodule "presence_mwi.so" >> loadmodule "presence_xml.so" >> loadmodule "presence_dialoginfo.so" >> loadmodule "presence_callinfo.so" >> loadmodule "pua.so" >> loadmodule "pua_dialoginfo.so" >> loadmodule "xcap.so" >> modparam("presence|xcap|pua","db_url","postgres://URI/opensips_prod01") >> modparam("presence","server_address","sip:proxy at PUBLIC IP:5082") >> modparam("presence", "notify_offline_body", 1) >> modparam("presence", "fallback2db", 1) >> modparam("presence", "clean_period", 30) >> modparam("presence", "mix_dialog_presence", 1) >> modparam("presence", "cluster_id", 1) >> modparam("presence", "cluster_sharing_tags", "A=active") >> modparam("presence", "cluster_federation_mode", 1) >> modparam("presence", "cluster_pres_events" ,"presence , dialog;sla") >> modparam("presence_xml", "force_active", 1) >> modparam("presence_xml", "pidf_manipulation", 1) >> modparam("pua_dialoginfo", "presence_server", "sip:proxy at PUBLIC >> IP:5082") >> >> >> volga629 >> >> >> On Thu, Jun 7, 2018 at 7:32 AM, Bogdan-Andrei Iancu >>  wrote: >>> Hi Slava, >>> >>> What is the presence clustering configuration you have here ? also, >>> what is the DB setup in regards to the cluster ? >>> >>> Also, did you start getting those errors only after enabling the >>> clustering support? have you run an opensips individual presence >>> node to see if you still get them ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> OpenSIPS Summit 2018 >>> http://www.opensips.org/events/Summit-2018Amsterdam >>> >>> On 06/06/2018 05:20 PM, volga629 at networklab.ca wrote: >>>> Hello Everyone, >>>> I am trying put togher 3 nodes ( 2 active 1 backup) presence >>>> cluster and log filled with messages regard cleanup timer. >>>> Any help thank you. >>>> >>>> opensips-2.4.1.b044f11ee-16.fc27.x86_64 >>>> >>>> >>>> Jun 6 09:15:12 sbc01 /usr/sbin/opensips[4584]: >>>> WARNING:core:timer_ticker: timer task >>>> already scheduled for 5201330 ms (now 5653910 ms), it may >>>> overlap.. >>>> Jun 6 09:15:13 sbc01 /usr/sbin/opensips[4584]: >>>> WARNING:core:timer_ticker: timer task >>>> already scheduled for 5201330 ms (now 5654900 ms), it may >>>> overlap.. >>>> >>>> >>>> Slava. >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> > > From pasandev at ymail.com Mon Jul 2 03:45:37 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Mon, 2 Jul 2018 07:45:37 +0000 (UTC) Subject: [OpenSIPS-Users] hep rtcp data dump to postgres References: <568416925.1297810.1530517537958.ref@mail.yahoo.com> Message-ID: <568416925.1297810.1530517537958@mail.yahoo.com> Hi Guys, I'm dumping my rtcp data which collect over hep interface of my opensips in to postgres and it doesn't store the json string in the database instead it pushes an encoded string which blows homer api when querying rtcp data. Same setup with mysql DB works perfectly fine, and data is saved as json string. It appears it stores the hep encoded msg into db without decoding. :/ Appreciate if you can look in to this. Let me know if you need any more info OS: debian stretch opensips : 2.3.3 /etc/opensips/opensips.cfg listen=hep_udp:10.3.1.150:9060 loadmodule "db_postgres.so" loadmodule "sipcapture.so" loadmodule "proto_hep.so" modparam("sipcapture", "db_url", "postgres://sipcapture:Apo31f24vna at 127.0.0.1/homer_data") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 1) modparam("sipcapture", "hep_route", "my_hep_route") modparam("sipcapture", "rtcp_table_name", "rtcp_capture") route[my_hep_route] {         ### hep_get([data type,] chunk_id, vendor_id_pvar, chunk_data_pvar)         ### data type is optional for most of the generic chunks         ### Full list here: http://www.opensips.org/html/docs/modules/2.2.x/sipcapture#hep_set_id         #Protocol ID         hep_get("11", "$var(vid)", "$var(data)");         $var(proto) = $(var(data){s.int});         #xlog("L_INFO", "Dumping $var(data) pkts via HEP");         #Logs Or Stats         if($var(proto) == 100 || $var(proto) == 99) {                 #hep_set("uint8", "2", , "1");                 hep_get("utf8-string", "0x11", "$var(vid)", "$var(correlation_id)");                 report_capture("logs_capture", "$var(correlation_id)", "1");                 exit;         } else if ( $var(data) == "RTCP") {                 hep_get("utf8-string", "0x11", "$var(vid)", "$var(correlation_id)");                 report_capture("rtcp_capture", "$var(correlation_id)", "5");                 exit;         }         if($var(data) == "SIP") {                 hep_resume_sip();         } else {                 exit;         } } debug log Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:sipcapture:db_sync_store: storing info... Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 121 chars, out: 121 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 12 chars, out: 12 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 13 chars, out: 13 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 9 chars, out: 9 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_submit_query: 0x7fcc64b62a98 PQsendQuery(insert into rtcp_capture (date,micro_ts,correlation_id,source_ip,source_port,destination_ip,destination_port,proto,family,type,node,msg ) values ('2018-07-02 17:42:39',1530517359466892,'0gQAAC8WAAACBAAALxYAALesqTJJyH2xO7MoiUXzBE8wM7k5guQDJr3dDDdK19yGM3EIud2vT1YL3QYtviLXYUkO7n48a2Qtwkv2C38+rns- at 210.87.44.32','210.87.44.32',58871,'10.239.238.68',15997,1,2,5,'homer01:2','\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')) -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Mon Jul 2 03:57:38 2018 From: callum.guy at x-on.co.uk (Callum Guy) Date: Mon, 2 Jul 2018 08:57:38 +0100 Subject: [OpenSIPS-Users] OpenSIPs + RTP High Availability Plans In-Reply-To: References: Message-ID: Hi All, Just wondering if this mail came through? Be very interested in hearing if anyone has any ideas regarding a "multi-master" rtp proxy service. To date the closest I have found is a Redis plugin for RTPEngine whih provides some concept of shared state however I wonder if there is a simpler approach out there ( https://github.com/ngvoice/rtpengine-redis-plugin) Best Regards, Callum On Wed, Jun 27, 2018 at 11:45 AM Callum Guy wrote: > Hi All, > > I am looking into updating my network to support live failover between two > OpenSIPs instances acting as an HA pair. We use a VIP address and want to > allow calls and audio streams to be maintained in the event of a single > node failure. At present the system works and allows new calls to be > processed when one node goes away however any active calls lose their media > streams. > > I’ve started exploring my options and am open to the idea of moving to use > RTPEngine or implementing some state storage in RTPProxy (if such a thing > exists) but I wanted to reach out to the community to get an experienced > opinion while I’m gathering ideas. > > I would very much appreciate any ideas/thoughts and have tried to explain > the current deployment below, just let me know if you need more details. > > *Current Setup* > Two identical CentOS 7.5 instances running OpenSIPs 2.3.3 and RTPProxy > 2.0.0. > The servers have individual IP addresses 172.18.0.111 and 172.18.0.112. > External traffic hits a single static public IP (*8.8.8.8* for > illustrative purposes) which forwards SIP+RTP traffic to a floating IP > address *172.18.0.110* (keepalived) which is owned by the current master > instance. > All traffic is then forwarded on to an internal IVR platform (FreeSWITCH) > so OpenSIPs is just acting as a pure B2BUA proxy which needs to work in > both directions (i.e. some calls originate from the IVR as well as carrier > network calls coming in). > > *OpenSIPs/RTPProxy Setup* > RTPProxy is served on a local socket > “unix:/var/run/rtpproxy/rtpproxy.sock”. > When an INVITE is being processed we run "rtpproxy_offer(“rfo”, [*8.8.8.8* > or *172.18.0.110* ])" in accordance with the request direction. > When any reply is received our onreply_route will invoke > "rtpproxy_offer(“rfo”, [*8.8.8.8* or *172.18.0.110* ])" again in > accordance with the request direction. > > Hopefully that is clear - the idea is to allow the remote systems to be > completely agnostic to which proxy handled the traffic. I presume that the > main issue is that the backup RTPProxy instance is unaware of the current > state and is not listening on the necessary ports and that this is > something which can be addressed? > > Finally if anyone has any other advice/articles regarding using OpenSIPs > in this way (dialog failover, using 2.4 features etc) then this would be > gratefully received! > > Many thanks, > > Callum > -- > Callum Guy > Head of Information Security > X-on > -- Callum Guy Head of Information Security X-on -- *0333 332 0000  |  www.x-on.co.uk   |   **     * X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From kurgan-rus at inbox.ru Mon Jul 2 04:36:10 2018 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Mon, 02 Jul 2018 11:36:10 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?DROUTING=3A_order_and_priority?= Message-ID: <1530520570.284974371@f253.i.mail.ru> Hi list! Last days I'm trying to master DROUTING module usage. This is a piece of script: if( $(rU{s.len}) > 3 ) { set_dlg_profile("callout"); if(!do_routing("", "W")) { sl_send_reply("500", "DRouting Error"); exit; } } ****************** I noticed, that "W" flag works only if I set weight for rules (in gwlist column of dr_rules). Firstly I set weight for carriers (in gwlist column), as it's also possible, according to documentation, but it haven't made any changes. sqlite> .schema dr_carriers CREATE TABLE dr_carriers ( id INTEGER PRIMARY KEY AUTOINCREMENT NOT NULL, carrierid CHAR(64) NOT NULL, gwlist CHAR(255) NOT NULL, flags INTEGER DEFAULT 0 NOT NULL, state INTEGER DEFAULT 0 NOT NULL, attrs CHAR(255) DEFAULT NULL, description CHAR(128) DEFAULT NULL, CONSTRAINT dr_carriers_dr_carrier_idx UNIQUE (carrierid) ); dr carriers sqlite_query: /usr/bin/sqlite3 /etc/opensips/db/opensips-no-ast2.db 'select * FROM dr_carriers ORDER BY carrierid; ' 1|carrierKTSmts|mts45_9,mts45_10,mts45_11,mts45_12|0|0||carrier MTS 2|carrierVOIP|c_asterisk_1,c_asterisk_2,c_asterisk_3|0|0||carrier VoIP ****************** sqlite> .schema dr_rules CREATE TABLE dr_rules ( ruleid INTEGER PRIMARY KEY AUTOINCREMENT NOT NULL, groupid CHAR(255) NOT NULL, prefix CHAR(64) NOT NULL, timerec CHAR(255) DEFAULT NULL, priority INTEGER DEFAULT 0 NOT NULL, routeid CHAR(255) DEFAULT NULL, gwlist CHAR(255) NOT NULL, attrs CHAR(255) DEFAULT NULL, description CHAR(128) DEFAULT NULL ); dr rules sqlite_query: /usr/bin/sqlite3 /etc/opensips/db/opensips-no-ast2.db 'select * FROM dr_rules ORDER BY groupid; ' 9|1|7912||0||#carrierKTSmts,#carrierVOIP||rule 1st, MTS through MTS gates and ast1 10|2|7912||0||#carrierVOIP=20,#carrierKTSmts=10||rule MTS through MTS gates and ast1 for groupid 2 Is it OK, or I do something wrong? ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ From kurgan-rus at inbox.ru Mon Jul 2 05:00:01 2018 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Mon, 02 Jul 2018 12:00:01 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?DROUTING=3A_order_and_priority?= In-Reply-To: <1530520570.284974371@f253.i.mail.ru> References: <1530520570.284974371@f253.i.mail.ru> Message-ID: <1530522001.436214024@f55.i.mail.ru> Now, the script remains as it was (no changes): if( $(rU{s.len}) > 3 ) { set_dlg_profile("callout"); if(!do_routing("", "W")) { sl_send_reply("500", "DRouting Error"); exit; } } But I edited DB - 1) changed the order of gw's in dr_carriers, 2) added weights there, 3) added a flag 0x1 to the table (use weight for sorting the list and not definition order). (then 'opensipsctl dr reload' of course) dr carriers sqlite_query: /usr/bin/sqlite3 /etc/opensips/db/opensips-no-ast2.db 'select * FROM dr_carriers ORDER BY carrierid; ' 1|carrierKTSmts|mts45_9,mts45_10,mts45_11,mts45_12|0|0||carrier MTS 2|carrierVOIP|c_asterisk_3=10,c_asterisk_2=20,c_asterisk_1=30|0x1|0||carrier VoIP According to this ^^^ the call should go to c_asterisk_1=30, - as it has the highest weight - as I set '0x1' flag in DB - as I still use "W" flag for 'do_routing' in the script. But no... the call goes to the c_asterisk_3=10, as it's just the first in the list. ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ From pasandev at ymail.com Mon Jul 2 06:20:22 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Mon, 2 Jul 2018 10:20:22 +0000 (UTC) Subject: [OpenSIPS-Users] hep rtcp data dump to postgres In-Reply-To: <568416925.1297810.1530517537958@mail.yahoo.com> References: <568416925.1297810.1530517537958.ref@mail.yahoo.com> <568416925.1297810.1530517537958@mail.yahoo.com> Message-ID: <1403999659.1350559.1530526822172@mail.yahoo.com> mysql debug log with same setup. Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:sipcapture:db_sync_store: storing info...Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:has_stmt_ctx: ctx found for rtcp_capture Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_do_prepared_query: conn=0x7f610eda28c8 (tail=140054837734000) MC=0x7f610eda2ae8 Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_do_prepared_query: set values for the statement run Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_val2bind: added val (0): len=40; type=12; is_null=0 Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_val2bind: added val (1): len=8; type=8; is_null=0 Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_val2bind: added val (2): len=121; type=254; is_null=0 Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_val2bind: added val (3): len=12; type=254; is_null=0 Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_val2bind: added val (4): len=4; type=3; is_null=0 Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_val2bind: added val (5): len=13; type=254; is_null=0 Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_val2bind: added val (6): len=4; type=3; is_null=0 Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_val2bind: added val (7): len=4; type=3; is_null=0 Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_val2bind: added val (8): len=4; type=3; is_null=0 Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_val2bind: added val (9): len=4; type=3; is_null=0 Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_val2bind: added val (10): len=9; type=254; is_null=0 Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_val2bind: added val (11): len=338; type=252; is_null=0 Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:db_mysql_do_prepared_query: doing BIND_PARAM in... Jul  2 20:15:48 voip2-homer /usr/sbin/opensips[19488]: DBG:db_mysql:mysql_raise_event: MySQL status has not changed: connected On Monday, 2 July 2018, 1:20:31 PM GMT+5:30, Pasan Meemaduma via Users wrote: Hi Guys, I'm dumping my rtcp data which collect over hep interface of my opensips in to postgres and it doesn't store the json string in the database instead it pushes an encoded string which blows homer api when querying rtcp data. Same setup with mysql DB works perfectly fine, and data is saved as json string. It appears it stores the hep encoded msg into db without decoding. :/ Appreciate if you can look in to this. Let me know if you need any more info OS: debian stretch opensips : 2.3.3 /etc/opensips/opensips.cfg listen=hep_udp:10.3.1.150:9060 loadmodule "db_postgres.so" loadmodule "sipcapture.so" loadmodule "proto_hep.so" modparam("sipcapture", "db_url", "postgres://sipcapture:Apo31f24vna at 127.0.0.1/homer_data") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 1) modparam("sipcapture", "hep_route", "my_hep_route") modparam("sipcapture", "rtcp_table_name", "rtcp_capture") route[my_hep_route] {         ### hep_get([data type,] chunk_id, vendor_id_pvar, chunk_data_pvar)         ### data type is optional for most of the generic chunks         ### Full list here: http://www.opensips.org/html/docs/modules/2.2.x/sipcapture#hep_set_id         #Protocol ID         hep_get("11", "$var(vid)", "$var(data)");         $var(proto) = $(var(data){s.int});         #xlog("L_INFO", "Dumping $var(data) pkts via HEP");         #Logs Or Stats         if($var(proto) == 100 || $var(proto) == 99) {                 #hep_set("uint8", "2", , "1");                 hep_get("utf8-string", "0x11", "$var(vid)", "$var(correlation_id)");                 report_capture("logs_capture", "$var(correlation_id)", "1");                 exit;         } else if ( $var(data) == "RTCP") {                 hep_get("utf8-string", "0x11", "$var(vid)", "$var(correlation_id)");                 report_capture("rtcp_capture", "$var(correlation_id)", "5");                 exit;         }         if($var(data) == "SIP") {                 hep_resume_sip();         } else {                 exit;         } } debug log Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:sipcapture:db_sync_store: storing info... Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 121 chars, out: 121 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 12 chars, out: 12 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 13 chars, out: 13 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 9 chars, out: 9 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_submit_query: 0x7fcc64b62a98 PQsendQuery(insert into rtcp_capture (date,micro_ts,correlation_id,source_ip,source_port,destination_ip,destination_port,proto,family,type,node,msg ) values ('2018-07-02 17:42:39',1530517359466892,'0gQAAC8WAAACBAAALxYAALesqTJJyH2xO7MoiUXzBE8wM7k5guQDJr3dDDdK19yGM3EIud2vT1YL3QYtviLXYUkO7n48a2Qtwkv2C38+rns- at 210.87.44.32','210.87.44.32',58871,'10.239.238.68',15997,1,2,5,'homer01:2','\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')) _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Mon Jul 2 22:41:44 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Tue, 3 Jul 2018 02:41:44 +0000 (UTC) Subject: [OpenSIPS-Users] hep rtcp data dump to postgres In-Reply-To: <568416925.1297810.1530517537958@mail.yahoo.com> References: <568416925.1297810.1530517537958.ref@mail.yahoo.com> <568416925.1297810.1530517537958@mail.yahoo.com> Message-ID: <2009243184.1852798.1530585704147@mail.yahoo.com> I figured its json data encoded in hex, so it appears DB_BLOB in OpenSIPS/opensips | | | | | | | | | | | OpenSIPS/opensips opensips - OpenSIPS is a GPL implementation of a multi-functionality SIP Server that targets to deliver a high-l... | | | On Monday, 2 July 2018, 1:20:31 PM GMT+5:30, Pasan Meemaduma via Users wrote: Hi Guys, I'm dumping my rtcp data which collect over hep interface of my opensips in to postgres and it doesn't store the json string in the database instead it pushes an encoded string which blows homer api when querying rtcp data. Same setup with mysql DB works perfectly fine, and data is saved as json string. It appears it stores the hep encoded msg into db without decoding. :/ Appreciate if you can look in to this. Let me know if you need any more info OS: debian stretch opensips : 2.3.3 /etc/opensips/opensips.cfg listen=hep_udp:10.3.1.150:9060 loadmodule "db_postgres.so" loadmodule "sipcapture.so" loadmodule "proto_hep.so" modparam("sipcapture", "db_url", "postgres://sipcapture:Apo31f24vna at 127.0.0.1/homer_data") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 1) modparam("sipcapture", "hep_route", "my_hep_route") modparam("sipcapture", "rtcp_table_name", "rtcp_capture") route[my_hep_route] {         ### hep_get([data type,] chunk_id, vendor_id_pvar, chunk_data_pvar)         ### data type is optional for most of the generic chunks         ### Full list here: http://www.opensips.org/html/docs/modules/2.2.x/sipcapture#hep_set_id         #Protocol ID         hep_get("11", "$var(vid)", "$var(data)");         $var(proto) = $(var(data){s.int});         #xlog("L_INFO", "Dumping $var(data) pkts via HEP");         #Logs Or Stats         if($var(proto) == 100 || $var(proto) == 99) {                 #hep_set("uint8", "2", , "1");                 hep_get("utf8-string", "0x11", "$var(vid)", "$var(correlation_id)");                 report_capture("logs_capture", "$var(correlation_id)", "1");                 exit;         } else if ( $var(data) == "RTCP") {                 hep_get("utf8-string", "0x11", "$var(vid)", "$var(correlation_id)");                 report_capture("rtcp_capture", "$var(correlation_id)", "5");                 exit;         }         if($var(data) == "SIP") {                 hep_resume_sip();         } else {                 exit;         } } debug log Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:sipcapture:db_sync_store: storing info... Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 121 chars, out: 121 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 12 chars, out: 12 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 13 chars, out: 13 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 9 chars, out: 9 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_submit_query: 0x7fcc64b62a98 PQsendQuery(insert into rtcp_capture (date,micro_ts,correlation_id,source_ip,source_port,destination_ip,destination_port,proto,family,type,node,msg ) values ('2018-07-02 17:42:39',1530517359466892,'0gQAAC8WAAACBAAALxYAALesqTJJyH2xO7MoiUXzBE8wM7k5guQDJr3dDDdK19yGM3EIud2vT1YL3QYtviLXYUkO7n48a2Qtwkv2C38+rns- at 210.87.44.32','210.87.44.32',58871,'10.239.238.68',15997,1,2,5,'homer01:2','\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')) _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Tue Jul 3 00:54:22 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Tue, 3 Jul 2018 04:54:22 +0000 (UTC) Subject: [OpenSIPS-Users] hep rtcp data dump to postgres In-Reply-To: <2009243184.1852798.1530585704147@mail.yahoo.com> References: <568416925.1297810.1530517537958.ref@mail.yahoo.com> <568416925.1297810.1530517537958@mail.yahoo.com> <2009243184.1852798.1530585704147@mail.yahoo.com> Message-ID: <1519312103.1868661.1530593662680@mail.yahoo.com> We were able to fix this with following change. Let me know if you need a pull request if you are happy with that change. --- opensips-2.3.4.orig/modules/sipcapture/sipcapture.c +++ opensips-2.3.4/modules/sipcapture/sipcapture.c @@ -5070,7 +5070,7 @@ static int report_capture(struct sip_msg          db_vals[10].val.str_val = capture_node;      }   -    db_vals[11].type = DB_BLOB; +    db_vals[11].type = DB_STR;          /* we can have other pyload than sip only for hepv3 */ On Tuesday, 3 July 2018, 8:16:50 AM GMT+5:30, Pasan Meemaduma via Users wrote: I figured its json data encoded in hex, so it appears DB_BLOB in OpenSIPS/opensips | | | | | | | | | | | OpenSIPS/opensips opensips - OpenSIPS is a GPL implementation of a multi-functionality SIP Server that targets to deliver a high-l... | | | On Monday, 2 July 2018, 1:20:31 PM GMT+5:30, Pasan Meemaduma via Users wrote: Hi Guys, I'm dumping my rtcp data which collect over hep interface of my opensips in to postgres and it doesn't store the json string in the database instead it pushes an encoded string which blows homer api when querying rtcp data. Same setup with mysql DB works perfectly fine, and data is saved as json string. It appears it stores the hep encoded msg into db without decoding. :/ Appreciate if you can look in to this. Let me know if you need any more info OS: debian stretch opensips : 2.3.3 /etc/opensips/opensips.cfg listen=hep_udp:10.3.1.150:9060 loadmodule "db_postgres.so" loadmodule "sipcapture.so" loadmodule "proto_hep.so" modparam("sipcapture", "db_url", "postgres://sipcapture:Apo31f24vna at 127.0.0.1/homer_data") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 1) modparam("sipcapture", "hep_route", "my_hep_route") modparam("sipcapture", "rtcp_table_name", "rtcp_capture") route[my_hep_route] {         ### hep_get([data type,] chunk_id, vendor_id_pvar, chunk_data_pvar)         ### data type is optional for most of the generic chunks         ### Full list here: http://www.opensips.org/html/docs/modules/2.2.x/sipcapture#hep_set_id         #Protocol ID         hep_get("11", "$var(vid)", "$var(data)");         $var(proto) = $(var(data){s.int});         #xlog("L_INFO", "Dumping $var(data) pkts via HEP");         #Logs Or Stats         if($var(proto) == 100 || $var(proto) == 99) {                 #hep_set("uint8", "2", , "1");                 hep_get("utf8-string", "0x11", "$var(vid)", "$var(correlation_id)");                 report_capture("logs_capture", "$var(correlation_id)", "1");                 exit;         } else if ( $var(data) == "RTCP") {                 hep_get("utf8-string", "0x11", "$var(vid)", "$var(correlation_id)");                 report_capture("rtcp_capture", "$var(correlation_id)", "5");                 exit;         }         if($var(data) == "SIP") {                 hep_resume_sip();         } else {                 exit;         } } debug log Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:sipcapture:db_sync_store: storing info... Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 121 chars, out: 121 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 12 chars, out: 12 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 13 chars, out: 13 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 9 chars, out: 9 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_submit_query: 0x7fcc64b62a98 PQsendQuery(insert into rtcp_capture (date,micro_ts,correlation_id,source_ip,source_port,destination_ip,destination_port,proto,family,type,node,msg ) values ('2018-07-02 17:42:39',1530517359466892,'0gQAAC8WAAACBAAALxYAALesqTJJyH2xO7MoiUXzBE8wM7k5guQDJr3dDDdK19yGM3EIud2vT1YL3QYtviLXYUkO7n48a2Qtwkv2C38+rns- at 210.87.44.32','210.87.44.32',58871,'10.239.238.68',15997,1,2,5,'homer01:2','\x7b2273737263223a313630353339313435302c2274797065223a3230302c227265706f72745f636f756e74223a312c2273656e6465725f696e666f726d6174696f6e223a7b226e74705f74696d657374616d705f736563223a2232303835393739303632222c226e74705f74696d657374616d705f75736563223a22363334343939222c227274705f74696d657374616d70223a2d36323238383439332c227061636b657473223a31383231362c226f6374657473223a323931343536307d2c227265706f72745f626c6f636b73223a5b7b22736f757263655f73737263223a3937353539343236322c226672616374696f6e5f6c6f7374223a302c227061636b6574735f6c6f7374223a312c22686967686573745f7365715f6e6f223a34373337312c2269615f6a6974746572223a302c226c7372223a2231343431363437323835222c22646c7372223a3130383932307d5d7d')) _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From ryandelgrosso at gmail.com Tue Jul 3 03:16:29 2018 From: ryandelgrosso at gmail.com (Ryan Delgrosso) Date: Tue, 3 Jul 2018 00:16:29 -0700 Subject: [OpenSIPS-Users] hep message parsing error question Message-ID: <3e081207-1575-574f-6ef5-b6652e357900@gmail.com> Greetings all, I have recently built a homer system using the V5 Opensips deploy script and am presently working on getting RTP from rtpengine into homer and keep encountering this error on the opensips side in the homer server ERROR:core:parse_method: invalid character { ERROR:core:parse_msg: message=<{ "sender_information":{"ntp_timestamp_sec":3739588863,"ntp_timestamp_usec":814699461,"octets":640,"rtp_timestamp":2263688717, "packets":4},"ssrc":303507491,"type":200,"report_count":0,"report_blocks":[],"sdes_ssrc":303507491,"sdes_report_count":1,"sdes_information": [ {"sdes_chunk_ssrc":303507491,"type":1,"text":"jYeeYPmX6A/YszS6"}] }> ERROR:core:receive_msg: Unable to parse msg received from [:] SIP packets from seem to come through just fine, and taking a network capture shows both streams begin with "HEP3" so both seem to be V3 streams. Ive also tried both UDP and TCP with the same results. I'm confident I'm missing something pretty basic here but combing through the modules docs and the homer docs hasn't yielded much Any help is much appreciated! -Ryan From pasandev at ymail.com Tue Jul 3 03:26:47 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Tue, 3 Jul 2018 07:26:47 +0000 (UTC) Subject: [OpenSIPS-Users] hep message parsing error question In-Reply-To: <3e081207-1575-574f-6ef5-b6652e357900@gmail.com> References: <3e081207-1575-574f-6ef5-b6652e357900@gmail.com> Message-ID: <1892556450.1932762.1530602807916@mail.yahoo.com> Hi Ryan, You need to handle RTCP pkt separately. you can't pass it to opensips because it expect a sip packet, not a json encoded object. You can use the following config to log your rtcp pkts to appropriate table. Change your HEP route as below, modparam("sipcapture", "rtcp_table_name", "rtcp_capture") route[my_hep_route] {         ### hep_get([data type,] chunk_id, vendor_id_pvar, chunk_data_pvar)         ### data type is optional for most of the generic chunks         ### Full list here: http://www.opensips.org/html/docs/modules/2.2.x/sipcapture#hep_set_id         #Protocol ID         hep_get("11", "$var(vid)", "$var(data)");         $var(proto) = $(var(data){s.int});         #xlog("L_INFO", "Dumping $var(data) pkts via HEP");         #Logs Or Stats         if($var(proto) == 100 || $var(proto) == 99) {                 #hep_set("uint8", "2", , "1");                 hep_get("utf8-string", "0x11", "$var(vid)", "$var(correlation_id)");                 report_capture("logs_capture", "$var(correlation_id)", "1");                 exit;         } else if ( $var(data) == "RTCP") {                 hep_get("utf8-string", "0x11", "$var(vid)", "$var(correlation_id)");                 report_capture("rtcp_capture", "$var(correlation_id)", "5");                 exit;         }         if($var(data) == "SIP") {                 hep_resume_sip();         } else {                 exit;         } } Hope this helps. On Tuesday, 3 July 2018, 12:47:12 PM GMT+5:30, Ryan Delgrosso wrote: Greetings all, I have recently built a homer system using the V5 Opensips deploy script and am presently working on getting RTP from rtpengine into homer and keep encountering this error on the opensips side in the homer server ERROR:core:parse_method: invalid character { ERROR:core:parse_msg: message=<{ "sender_information":{"ntp_timestamp_sec":3739588863,"ntp_timestamp_usec":814699461,"octets":640,"rtp_timestamp":2263688717, "packets":4},"ssrc":303507491,"type":200,"report_count":0,"report_blocks":[],"sdes_ssrc":303507491,"sdes_report_count":1,"sdes_information": [ {"sdes_chunk_ssrc":303507491,"type":1,"text":"jYeeYPmX6A/YszS6"}] }> ERROR:core:receive_msg: Unable to parse msg received from [:] SIP packets from seem to come through just fine, and taking a network capture shows both streams begin with "HEP3" so both seem to be V3 streams. Ive also tried both UDP and TCP with the same results. I'm confident I'm missing something pretty basic here but combing through the modules docs and the homer docs hasn't yielded much Any help is much appreciated! -Ryan _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue Jul 3 09:00:38 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 3 Jul 2018 16:00:38 +0300 Subject: [OpenSIPS-Users] [RELEASES] OpenSIPS Releases Lifetime Planning Message-ID: Hello everyone! We have just updated on our website[1] the lifetime of all available OpenSIPS releases. Here is a summary: * OpenSIPS 2.3 - maintained until 26th of October 2018 * OpenSIPS 2.2 LTS - supported until 27th of May 2019 * OpenSIPS 2.4 LTS - supported until 24th of May 2021 We recommend all OpenSIPS 2.3 users to upgrade their installs to OpenSIPS 2.4 as soon as posible. [1] http://www.opensips.org/About/AvailableVersions Best regards, -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From yuval.dinari at vonage.com Wed Jul 4 03:46:43 2018 From: yuval.dinari at vonage.com (Yuval Dinari) Date: Wed, 4 Jul 2018 10:46:43 +0300 Subject: [OpenSIPS-Users] Retransmissions Message-ID: Hi, 1. How can I control the amount/time window of retransmissions? 2. I see a behavior in which opensips sends an INVITE to a non responsive client. After a few retransmissions it sends a 408 to the originator of the INVITE, and then sends some more retransmissions of the INVITE. I don't understand why it send INVITE-s after the 408. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Wed Jul 4 04:36:49 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Wed, 4 Jul 2018 08:36:49 +0000 (UTC) Subject: [OpenSIPS-Users] Retransmissions In-Reply-To: References: Message-ID: <1585133444.2581418.1530693409713@mail.yahoo.com> Hi Yuval, SIP Timers related functions can be found in TM (transaction) module http://www.opensips.org/html/docs/modules/2.4.x/tm.html hard to say whats causing retransmission after a final reply without knowing your config. | | | | On Wednesday, 4 July 2018, 1:17:41 PM GMT+5:30, Yuval Dinari via Users wrote: Hi,1. How can I control the amount/time window of retransmissions?2. I see a behavior in which opensips sends an INVITE to a non responsive client. After a few retransmissions it sends a 408 to the originator of the INVITE, and then sends some more retransmissions of the INVITE. I don't understand why it send INVITE-s after the 408. Thanks_______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From yuval.dinari at vonage.com Wed Jul 4 11:43:35 2018 From: yuval.dinari at vonage.com (Yuval Dinari) Date: Wed, 4 Jul 2018 18:43:35 +0300 Subject: [OpenSIPS-Users] Retransmissions In-Reply-To: <1585133444.2581418.1530693409713@mail.yahoo.com> References: <1585133444.2581418.1530693409713@mail.yahoo.com> Message-ID: I set fr_timeout parameter of tm module to 5, so after 5 seconds opensips generates a 408 response, but keeps retransmitting the INVITE even after that. Apart from that the handling of INVITE is pretty straight forward: Do lookup to find the destination (callee), and if found call t_relay(). Thanks On Wed, Jul 4, 2018 at 11:40 AM Pasan Meemaduma wrote: > Hi Yuval, > SIP Timers related functions can be found in TM (transaction) module > http://www.opensips.org/html/docs/modules/2.4.x/tm.html > > hard to say whats causing retransmission after a final reply without > knowing your config. > > > > > > > > > On Wednesday, 4 July 2018, 1:17:41 PM GMT+5:30, Yuval Dinari via Users < > users at lists.opensips.org> wrote: > > > Hi, > 1. How can I control the amount/time window of retransmissions? > 2. I see a behavior in which opensips sends an INVITE to a non responsive > client. After a few retransmissions it sends a 408 to the originator of the > INVITE, and then sends some more retransmissions of the INVITE. I don't > understand why it send INVITE-s after the 408. > > Thanks > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Wed Jul 4 11:56:47 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Wed, 4 Jul 2018 15:56:47 +0000 (UTC) Subject: [OpenSIPS-Users] Retransmissions In-Reply-To: References: <1585133444.2581418.1530693409713@mail.yahoo.com> Message-ID: <1484883132.2696688.1530719807702@mail.yahoo.com> which version of opensips are you using ? I'm not aware of retransmission after a final reply or timeout. if you can provide a config to replicate the issue or logs of opensips we might able to check more On Wednesday, 4 July 2018, 9:13:49 PM GMT+5:30, Yuval Dinari wrote: I set fr_timeout parameter of tm module to 5, so after 5 seconds opensips generates a 408 response, but keeps retransmitting the INVITE even after that.Apart from that the handling of INVITE is pretty straight forward:Do lookup to find the destination (callee), and if found call t_relay().Thanks On Wed, Jul 4, 2018 at 11:40 AM Pasan Meemaduma wrote: Hi Yuval, SIP Timers related functions can be found in TM (transaction) module http://www.opensips.org/html/docs/modules/2.4.x/tm.html hard to say whats causing retransmission after a final reply without knowing your config. | | | | On Wednesday, 4 July 2018, 1:17:41 PM GMT+5:30, Yuval Dinari via Users wrote: Hi,1. How can I control the amount/time window of retransmissions?2. I see a behavior in which opensips sends an INVITE to a non responsive client. After a few retransmissions it sends a 408 to the originator of the INVITE, and then sends some more retransmissions of the INVITE. I don't understand why it send INVITE-s after the 408. Thanks_______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosenberg11219 at gmail.com Thu Jul 5 02:43:19 2018 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Thu, 5 Jul 2018 09:43:19 +0300 Subject: [OpenSIPS-Users] location table in MySQL when using full-sharing-cluster Message-ID: Hi I have the working_mode_preset parameter in the usrloc module set to full-sharing-cluster, is there anyway to have the location table written to the MySQL database too? I need to pull information from the database. Basically the database should be for informational purposes, the restart_persistency should still be sync-from-cluster and the sql_write_mode should be write-back. A nice option would be if It can also be used as a backup for restart_persistency if the entire cluster crashes at once. I assume I can get this done with NoSQL but that would mean that I need to rewrite a lot of stuff that was written for MySQL S. Rosenberg From liviu at opensips.org Thu Jul 5 03:30:42 2018 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 5 Jul 2018 10:30:42 +0300 Subject: [OpenSIPS-Users] location table in MySQL when using full-sharing-cluster In-Reply-To: References: Message-ID: Hi Schneur, Yes, this is possible.  Unset the "working_mode_preset" and individually set the "cluster_mode", "restart_persistency" and "sql_write_mode" so you achieve this effect. Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 05.07.2018 09:43, Schneur Rosenberg wrote: > Hi I have the working_mode_preset parameter in the usrloc module set > to full-sharing-cluster, is there anyway to have the location table > written to the MySQL database too? I need to pull information from > the database. > > Basically the database should be for informational purposes, the > restart_persistency should still be sync-from-cluster and the > sql_write_mode should be write-back. > > A nice option would be if It can also be used as a backup for > restart_persistency if the entire cluster crashes at once. > > I assume I can get this done with NoSQL but that would mean that I > need to rewrite a lot of stuff that was written for MySQL > > S. Rosenberg > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From liviu at opensips.org Thu Jul 5 04:29:30 2018 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 5 Jul 2018 11:29:30 +0300 Subject: [OpenSIPS-Users] location table in MySQL when using full-sharing-cluster In-Reply-To: References: Message-ID: <05a51707-7431-43af-8a38-6a01fdf8d7ad@opensips.org> Sorry, I didn't fully read your request.  The "restart_persistency" parameter is a selector: either A) or B). What you want is something like: "perform A). if it fails, fall back to B)".  And this is, indeed, not possible as of now. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 05.07.2018 10:30, Liviu Chircu wrote: > Hi Schneur, > > Yes, this is possible.  Unset the "working_mode_preset" and > individually set the "cluster_mode", "restart_persistency" and > "sql_write_mode" so you achieve this effect. > > Best regards, > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 05.07.2018 09:43, Schneur Rosenberg wrote: >> Hi I have the working_mode_preset parameter in the usrloc module set >> to full-sharing-cluster, is there anyway to have the location table >> written to the MySQL database too? I need  to pull information from >> the database. >> >> Basically the database should be for informational purposes, the >> restart_persistency should still be sync-from-cluster and the >> sql_write_mode should be write-back. >> >> A nice option would be if It can also be used as a backup for >> restart_persistency if the entire cluster crashes at once. >> >> I assume I can get this done with NoSQL but that would mean that I >> need to rewrite a lot of stuff that was written for MySQL >> >> S. Rosenberg >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From pasandev at ymail.com Thu Jul 5 04:28:51 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Thu, 5 Jul 2018 08:28:51 +0000 (UTC) Subject: [OpenSIPS-Users] hep rtcp data dump to postgres In-Reply-To: <1519312103.1868661.1530593662680@mail.yahoo.com> References: <568416925.1297810.1530517537958.ref@mail.yahoo.com> <568416925.1297810.1530517537958@mail.yahoo.com> <2009243184.1852798.1530585704147@mail.yahoo.com> <1519312103.1868661.1530593662680@mail.yahoo.com> Message-ID: <1265949755.2948878.1530779331033@mail.yahoo.com> Hi Guys, Would you please confirm this change ? On Tuesday, 3 July 2018, 10:29:05 AM GMT+5:30, Pasan Meemaduma via Users wrote: We were able to fix this with following change. Let me know if you need a pull request if you are happy with that change. --- opensips-2.3.4.orig/modules/sipcapture/sipcapture.c +++ opensips-2.3.4/modules/sipcapture/sipcapture.c @@ -5070,7 +5070,7 @@ static int report_capture(struct sip_msg          db_vals[10].val.str_val = capture_node;      }   -    db_vals[11].type = DB_BLOB; +    db_vals[11].type = DB_STR;          /* we can have other pyload than sip only for hepv3 */ On Tuesday, 3 July 2018, 8:16:50 AM GMT+5:30, Pasan Meemaduma via Users wrote: I figured its json data encoded in hex, so it appears DB_BLOB in OpenSIPS/opensips | | | | | | | | | | | OpenSIPS/opensips opensips - OpenSIPS is a GPL implementation of a multi-functionality SIP Server that targets to deliver a high-l... | | | On Monday, 2 July 2018, 1:20:31 PM GMT+5:30, Pasan Meemaduma via Users wrote: Hi Guys, I'm dumping my rtcp data which collect over hep interface of my opensips in to postgres and it doesn't store the json string in the database instead it pushes an encoded string which blows homer api when querying rtcp data. Same setup with mysql DB works perfectly fine, and data is saved as json string. It appears it stores the hep encoded msg into db without decoding. :/ Appreciate if you can look in to this. Let me know if you need any more info OS: debian stretch opensips : 2.3.3 /etc/opensips/opensips.cfg listen=hep_udp:10.3.1.150:9060 loadmodule "db_postgres.so" loadmodule "sipcapture.so" loadmodule "proto_hep.so" modparam("sipcapture", "db_url", "postgres://sipcapture:Apo31f24vna at 127.0.0.1/homer_data") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 1) modparam("sipcapture", "hep_route", "my_hep_route") modparam("sipcapture", "rtcp_table_name", "rtcp_capture") route[my_hep_route] {         ### hep_get([data type,] chunk_id, vendor_id_pvar, chunk_data_pvar)         ### data type is optional for most of the generic chunks         ### Full list here: http://www.opensips.org/html/docs/modules/2.2.x/sipcapture#hep_set_id         #Protocol ID         hep_get("11", "$var(vid)", "$var(data)");         $var(proto) = $(var(data){s.int});         #xlog("L_INFO", "Dumping $var(data) pkts via HEP");         #Logs Or Stats         if($var(proto) == 100 || $var(proto) == 99) {                 #hep_set("uint8", "2", , "1");                 hep_get("utf8-string", "0x11", "$var(vid)", "$var(correlation_id)");                 report_capture("logs_capture", "$var(correlation_id)", "1");                 exit;         } else if ( $var(data) == "RTCP") {                 hep_get("utf8-string", "0x11", "$var(vid)", "$var(correlation_id)");                 report_capture("rtcp_capture", "$var(correlation_id)", "5");                 exit;         }         if($var(data) == "SIP") {                 hep_resume_sip();         } else {                 exit;         } } debug log Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:sipcapture:db_sync_store: storing info... Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 121 chars, out: 121 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 12 chars, out: 12 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 13 chars, out: 13 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 9 chars, out: 9 chars Jul  2 17:42:39 voip2-homer /usr/sbin/opensips[17258]: DBG:db_postgres:db_postgres_submit_query: 0x7fcc64b62a98 PQsendQuery(insert into rtcp_capture (date,micro_ts,correlation_id,source_ip,source_port,destination_ip,destination_port,proto,family,type,node,msg ) values ('2018-07-02 17:42:39',1530517359466892,'0gQAAC8WAAACBAAALxYAALesqTJJyH2xO7MoiUXzBE8wM7k5guQDJr3dDDdK19yGM3EIud2vT1YL3QYtviLXYUkO7n48a2Qtwkv2C38+rns- at 210.87.44.32','210.87.44.32',58871,'10.239.238.68',15997,1,2,5,'homer01:2','\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')) _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosenberg11219 at gmail.com Thu Jul 5 04:47:31 2018 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Thu, 5 Jul 2018 11:47:31 +0300 Subject: [OpenSIPS-Users] location table in MySQL when using full-sharing-cluster In-Reply-To: <05a51707-7431-43af-8a38-6a01fdf8d7ad@opensips.org> References: <05a51707-7431-43af-8a38-6a01fdf8d7ad@opensips.org> Message-ID: Thanks liviu, the failover was just something nice to have, I can live without it, thanks again. On Thu, Jul 5, 2018, 11:32 AM Liviu Chircu wrote: > Sorry, I didn't fully read your request. The "restart_persistency" > parameter is a selector: either A) or B). What you want is something > like: "perform A). if it fails, fall back to B)". And this is, indeed, > not possible as of now. > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 05.07.2018 10:30, Liviu Chircu wrote: > > Hi Schneur, > > > > Yes, this is possible. Unset the "working_mode_preset" and > > individually set the "cluster_mode", "restart_persistency" and > > "sql_write_mode" so you achieve this effect. > > > > Best regards, > > > > Liviu Chircu > > OpenSIPS Developer > > http://www.opensips-solutions.com > > > > On 05.07.2018 09:43, Schneur Rosenberg wrote: > >> Hi I have the working_mode_preset parameter in the usrloc module set > >> to full-sharing-cluster, is there anyway to have the location table > >> written to the MySQL database too? I need to pull information from > >> the database. > >> > >> Basically the database should be for informational purposes, the > >> restart_persistency should still be sync-from-cluster and the > >> sql_write_mode should be write-back. > >> > >> A nice option would be if It can also be used as a backup for > >> restart_persistency if the entire cluster crashes at once. > >> > >> I assume I can get this done with NoSQL but that would mean that I > >> need to rewrite a lot of stuff that was written for MySQL > >> > >> S. Rosenberg > >> > >> _______________________________________________ > >> Users mailing list > >> Users at lists.opensips.org > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosenberg11219 at gmail.com Thu Jul 5 09:31:34 2018 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Thu, 5 Jul 2018 16:31:34 +0300 Subject: [OpenSIPS-Users] location table in MySQL when using full-sharing-cluster In-Reply-To: References: <05a51707-7431-43af-8a38-6a01fdf8d7ad@opensips.org> Message-ID: Liviu, can you help me out, I was playing around with the settings, by doing a ngrep I see the binary stuff going between servers, but the second server does not actually load the registrations in memory. When I do a restart on the second server it loads from DB into memory, but as soon as it expires it disappears, even though it received from the binary re-registrations from the cluster. Here is what I have set now. modparam("usrloc", "location_cluster", 1) modparam("usrloc", "skip_replicated_db_ops", 1) modparam("usrloc", "cluster_mode", "full-sharing") modparam("usrloc", "location_cluster", 1) modparam("usrloc", "restart_persistency", "load-from-sql") #modparam("usrloc", "restart_persistency", "sync-from-cluster") modparam("usrloc", "sql_write_mode", "write-back") modparam("usrloc", "db_mode", 2) #is this the same as the previous line? On Thu, Jul 5, 2018 at 11:47 AM, Schneur Rosenberg wrote: > Thanks liviu, the failover was just something nice to have, I can live > without it, thanks again. > > > On Thu, Jul 5, 2018, 11:32 AM Liviu Chircu wrote: >> >> Sorry, I didn't fully read your request. The "restart_persistency" >> parameter is a selector: either A) or B). What you want is something >> like: "perform A). if it fails, fall back to B)". And this is, indeed, >> not possible as of now. >> >> Liviu Chircu >> OpenSIPS Developer >> http://www.opensips-solutions.com >> >> On 05.07.2018 10:30, Liviu Chircu wrote: >> > Hi Schneur, >> > >> > Yes, this is possible. Unset the "working_mode_preset" and >> > individually set the "cluster_mode", "restart_persistency" and >> > "sql_write_mode" so you achieve this effect. >> > >> > Best regards, >> > >> > Liviu Chircu >> > OpenSIPS Developer >> > http://www.opensips-solutions.com >> > >> > On 05.07.2018 09:43, Schneur Rosenberg wrote: >> >> Hi I have the working_mode_preset parameter in the usrloc module set >> >> to full-sharing-cluster, is there anyway to have the location table >> >> written to the MySQL database too? I need to pull information from >> >> the database. >> >> >> >> Basically the database should be for informational purposes, the >> >> restart_persistency should still be sync-from-cluster and the >> >> sql_write_mode should be write-back. >> >> >> >> A nice option would be if It can also be used as a backup for >> >> restart_persistency if the entire cluster crashes at once. >> >> >> >> I assume I can get this done with NoSQL but that would mean that I >> >> need to rewrite a lot of stuff that was written for MySQL >> >> >> >> S. Rosenberg >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > >> > >> > _______________________________________________ >> > Users mailing list >> > Users at lists.opensips.org >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users From liviu at opensips.org Thu Jul 5 10:43:20 2018 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 5 Jul 2018 17:43:20 +0300 Subject: [OpenSIPS-Users] location table in MySQL when using full-sharing-cluster In-Reply-To: References: <05a51707-7431-43af-8a38-6a01fdf8d7ad@opensips.org> Message-ID: <5c75de97-155d-65ec-4457-922320a74c4b@opensips.org> Please don't set the "db_mode" [1] - it acts as a "working_mode_preset", overriding any other settings and putting your box into "single-instance-write-back".  Maybe I should emphasize this behavior even further both in the docs and logs. Best regards, [1]: http://www.opensips.org/html/docs/modules/2.4.x/usrloc.html#param_db_mode Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 05.07.2018 16:31, Schneur Rosenberg wrote: > Liviu, can you help me out, I was playing around with the settings, by > doing a ngrep I see the binary stuff going between servers, but the > second server does not actually load the registrations in memory. > > When I do a restart on the second server it loads from DB into memory, > but as soon as it expires it disappears, even though it received from > the binary re-registrations from the cluster. > > Here is what I have set now. > modparam("usrloc", "location_cluster", 1) > modparam("usrloc", "skip_replicated_db_ops", 1) > modparam("usrloc", "cluster_mode", "full-sharing") > modparam("usrloc", "location_cluster", 1) > modparam("usrloc", "restart_persistency", "load-from-sql") > #modparam("usrloc", "restart_persistency", "sync-from-cluster") > modparam("usrloc", "sql_write_mode", "write-back") > modparam("usrloc", "db_mode", 2) #is this the same as the previous line? > > On Thu, Jul 5, 2018 at 11:47 AM, Schneur Rosenberg > wrote: >> Thanks liviu, the failover was just something nice to have, I can live >> without it, thanks again. >> >> >> On Thu, Jul 5, 2018, 11:32 AM Liviu Chircu wrote: >>> Sorry, I didn't fully read your request. The "restart_persistency" >>> parameter is a selector: either A) or B). What you want is something >>> like: "perform A). if it fails, fall back to B)". And this is, indeed, >>> not possible as of now. >>> >>> Liviu Chircu >>> OpenSIPS Developer >>> http://www.opensips-solutions.com >>> >>> On 05.07.2018 10:30, Liviu Chircu wrote: >>>> Hi Schneur, >>>> >>>> Yes, this is possible. Unset the "working_mode_preset" and >>>> individually set the "cluster_mode", "restart_persistency" and >>>> "sql_write_mode" so you achieve this effect. >>>> >>>> Best regards, >>>> >>>> Liviu Chircu >>>> OpenSIPS Developer >>>> http://www.opensips-solutions.com >>>> >>>> On 05.07.2018 09:43, Schneur Rosenberg wrote: >>>>> Hi I have the working_mode_preset parameter in the usrloc module set >>>>> to full-sharing-cluster, is there anyway to have the location table >>>>> written to the MySQL database too? I need to pull information from >>>>> the database. >>>>> >>>>> Basically the database should be for informational purposes, the >>>>> restart_persistency should still be sync-from-cluster and the >>>>> sql_write_mode should be write-back. >>>>> >>>>> A nice option would be if It can also be used as a backup for >>>>> restart_persistency if the entire cluster crashes at once. >>>>> >>>>> I assume I can get this done with NoSQL but that would mean that I >>>>> need to rewrite a lot of stuff that was written for MySQL >>>>> >>>>> S. Rosenberg >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From rosenberg11219 at gmail.com Thu Jul 5 11:08:11 2018 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Thu, 5 Jul 2018 18:08:11 +0300 Subject: [OpenSIPS-Users] location table in MySQL when using full-sharing-cluster In-Reply-To: <5c75de97-155d-65ec-4457-922320a74c4b@opensips.org> References: <05a51707-7431-43af-8a38-6a01fdf8d7ad@opensips.org> <5c75de97-155d-65ec-4457-922320a74c4b@opensips.org> Message-ID: Thanks, that was the issue!!! On Thu, Jul 5, 2018 at 5:43 PM, Liviu Chircu wrote: > Please don't set the "db_mode" [1] - it acts as a "working_mode_preset", > overriding any other settings and putting your box into > "single-instance-write-back". Maybe I should emphasize this behavior even > further both in the docs and logs. > > Best regards, > > [1]: > http://www.opensips.org/html/docs/modules/2.4.x/usrloc.html#param_db_mode > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 05.07.2018 16:31, Schneur Rosenberg wrote: >> >> Liviu, can you help me out, I was playing around with the settings, by >> doing a ngrep I see the binary stuff going between servers, but the >> second server does not actually load the registrations in memory. >> >> When I do a restart on the second server it loads from DB into memory, >> but as soon as it expires it disappears, even though it received from >> the binary re-registrations from the cluster. >> >> Here is what I have set now. >> modparam("usrloc", "location_cluster", 1) >> modparam("usrloc", "skip_replicated_db_ops", 1) >> modparam("usrloc", "cluster_mode", "full-sharing") >> modparam("usrloc", "location_cluster", 1) >> modparam("usrloc", "restart_persistency", "load-from-sql") >> #modparam("usrloc", "restart_persistency", "sync-from-cluster") >> modparam("usrloc", "sql_write_mode", "write-back") >> modparam("usrloc", "db_mode", 2) #is this the same as the previous line? >> >> On Thu, Jul 5, 2018 at 11:47 AM, Schneur Rosenberg >> wrote: >>> >>> Thanks liviu, the failover was just something nice to have, I can live >>> without it, thanks again. >>> >>> >>> On Thu, Jul 5, 2018, 11:32 AM Liviu Chircu wrote: >>>> >>>> Sorry, I didn't fully read your request. The "restart_persistency" >>>> parameter is a selector: either A) or B). What you want is something >>>> like: "perform A). if it fails, fall back to B)". And this is, indeed, >>>> not possible as of now. >>>> >>>> Liviu Chircu >>>> OpenSIPS Developer >>>> http://www.opensips-solutions.com >>>> >>>> On 05.07.2018 10:30, Liviu Chircu wrote: >>>>> >>>>> Hi Schneur, >>>>> >>>>> Yes, this is possible. Unset the "working_mode_preset" and >>>>> individually set the "cluster_mode", "restart_persistency" and >>>>> "sql_write_mode" so you achieve this effect. >>>>> >>>>> Best regards, >>>>> >>>>> Liviu Chircu >>>>> OpenSIPS Developer >>>>> http://www.opensips-solutions.com >>>>> >>>>> On 05.07.2018 09:43, Schneur Rosenberg wrote: >>>>>> >>>>>> Hi I have the working_mode_preset parameter in the usrloc module set >>>>>> to full-sharing-cluster, is there anyway to have the location table >>>>>> written to the MySQL database too? I need to pull information from >>>>>> the database. >>>>>> >>>>>> Basically the database should be for informational purposes, the >>>>>> restart_persistency should still be sync-from-cluster and the >>>>>> sql_write_mode should be write-back. >>>>>> >>>>>> A nice option would be if It can also be used as a backup for >>>>>> restart_persistency if the entire cluster crashes at once. >>>>>> >>>>>> I assume I can get this done with NoSQL but that would mean that I >>>>>> need to rewrite a lot of stuff that was written for MySQL >>>>>> >>>>>> S. Rosenberg >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users at lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From k.galinurov at gmail.com Fri Jul 6 04:04:21 2018 From: k.galinurov at gmail.com (Kirill Galinurov) Date: Fri, 6 Jul 2018 11:04:21 +0300 Subject: [OpenSIPS-Users] drouting module in opensips 2.4 Message-ID: Hi all I migrate from 2.3 to in 2.4 on Centos 7.4. OpenSIPS 2.3 starts without going to background, while drouting module fetches all rules from DB into memory. OpenSIPS 2.4 goes to background immediately, and fetches rules only after that. If you try opensipsctl fifo dr_number_routing mts 0 9195643210 command during this moment it restart dr_reload. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Fri Jul 6 05:57:07 2018 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Fri, 06 Jul 2018 11:57:07 +0200 Subject: [OpenSIPS-Users] RTPengine provisioning by db Message-ID: Hi all, I’m trying to provionning rtpengine via mysql db, but i can’t add a new proxy via opensipctl My parameters are : ###################################################################### #### RTPENGINE module ###################################################################### #modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 udp:10.207.201.20:2223") i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") modparam("rtpengine", "db_url", "mysql://opensips:XXXXXXXX at ZZZZZZZZZZZ/opensips_lb_did_2_4") modparam("rtpengine", "db_table", "rtpproxy_sockets") modparam("rtpengine", "socket_column", "rtpproxy_sock") modparam("rtpengine", "set_column", "set_id") when i run : opensipsctl fifo rtpengine_show i received this error : ERROR: /tmp/opensips_fifo does not exist ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")' in your config ERROR: and also have loaded the mi_fifo module. When i use « rtpengine_sock », it works version: opensips 2.4.1 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. git revision: 5d042cffc main.c compiled on 09:17:20 Jul  6 2018 with gcc 4.7 Someone can help me ? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Fri Jul 6 06:18:01 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Fri, 6 Jul 2018 10:18:01 +0000 (UTC) Subject: [OpenSIPS-Users] RTPengine provisioning by db In-Reply-To: References: Message-ID: <342869488.15527.1530872281196@mail.yahoo.com> Hi Alain, do you have following config lines in your opensips.cfg file which enables the fifo management interface ? loadmodule "mi_fifo.so" # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") According to the error you got it doesn't seems to exists. On Friday, 6 July 2018, 3:27:51 PM GMT+5:30, Alain Bieuzent wrote: Hi all,   I’m trying to provionning rtpengine via mysql db, but i can’t add a new proxy via opensipctl   My parameters are :   ###################################################################### #### RTPENGINE module ###################################################################### #modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 udp:10.207.201.20:2223") i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") modparam("rtpengine", "db_url", "mysql://opensips:XXXXXXXX at ZZZZZZZZZZZ/opensips_lb_did_2_4") modparam("rtpengine", "db_table", "rtpproxy_sockets") modparam("rtpengine", "socket_column", "rtpproxy_sock") modparam("rtpengine", "set_column", "set_id")   when i run : opensipsctl fifo rtpengine_show i received this error : ERROR: /tmp/opensips_fifo does not exist ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")' in your config ERROR: and also have loaded the mi_fifo module.   When i use « rtpengine_sock », it works   version: opensips 2.4.1 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. git revision: 5d042cffc main.c compiled on 09:17:20 Jul  6 2018 with gcc 4.7   Someone can help me ?   thanks _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Fri Jul 6 08:05:28 2018 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Fri, 06 Jul 2018 14:05:28 +0200 Subject: [OpenSIPS-Users] RTPengine provisioning by db In-Reply-To: <342869488.15527.1530872281196@mail.yahoo.com> References: <342869488.15527.1530872281196@mail.yahoo.com> Message-ID: Hi Pasan, Thanks for the reply but yes of course I have mi_fifo module load and configure. I’m using also dispatcher with a db provisioning and without the db parameters for the rtpengine module I can manage the dispatcher entry via opensipsctl fi ds_XXXX with the the db parameters for the rtpengine module, command opensipsctl fifo ds_list provide the same error about mi_fifo. thanks De : Users au nom de Pasan Meemaduma via Users Répondre à : Pasan Meemaduma , OpenSIPS users mailling list Date : vendredi 6 juillet 2018 à 12:23 À : OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] RTPengine provisioning by db Hi Alain, do you have following config lines in your opensips.cfg file which enables the fifo management interface ? loadmodule "mi_fifo.so" # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") According to the error you got it doesn't seems to exists. On Friday, 6 July 2018, 3:27:51 PM GMT+5:30, Alain Bieuzent wrote: Hi all, I’m trying to provionning rtpengine via mysql db, but i can’t add a new proxy via opensipctl My parameters are : ###################################################################### #### RTPENGINE module ###################################################################### #modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 udp:10.207.201.20:2223") i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") modparam("rtpengine", "db_url", "mysql://opensips:XXXXXXXX at ZZZZZZZZZZZ/opensips_lb_did_2_4") modparam("rtpengine", "db_table", "rtpproxy_sockets") modparam("rtpengine", "socket_column", "rtpproxy_sock") modparam("rtpengine", "set_column", "set_id") when i run : opensipsctl fifo rtpengine_show i received this error : ERROR: /tmp/opensips_fifo does not exist ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")' in your config ERROR: and also have loaded the mi_fifo module. When i use « rtpengine_sock », it works version: opensips 2.4.1 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. git revision: 5d042cffc main.c compiled on 09:17:20 Jul 6 2018 with gcc 4.7 Someone can help me ? thanks _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Fri Jul 6 10:58:29 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Fri, 6 Jul 2018 14:58:29 +0000 (UTC) Subject: [OpenSIPS-Users] RTPengine provisioning by db In-Reply-To: References: <342869488.15527.1530872281196@mail.yahoo.com> Message-ID: <650397766.104778.1530889109323@mail.yahoo.com> ok , when you set the dbmode does fifo file exists in /tmp/opensips_fifo because according to the error, It appears your fifo file hasn't init after opensips start. if so one of dev from opensips might able shed some light here :) On Friday, 6 July 2018, 5:35:33 PM GMT+5:30, Alain Bieuzent wrote: #yiv2925365401 #yiv2925365401 -- _filtered #yiv2925365401 {panose-1:2 4 5 3 5 4 6 3 2 4;} _filtered #yiv2925365401 {font-family:Calibri;panose-1:2 15 5 2 2 2 4 3 2 4;} _filtered #yiv2925365401 {panose-1:2 0 5 3 0 0 0 2 0 4;}#yiv2925365401 #yiv2925365401 p.yiv2925365401MsoNormal, #yiv2925365401 li.yiv2925365401MsoNormal, #yiv2925365401 div.yiv2925365401MsoNormal {margin:0cm;margin-bottom:.0001pt;font-size:11.0pt;font-family:sans-serif;}#yiv2925365401 a:link, #yiv2925365401 span.yiv2925365401MsoHyperlink {color:blue;text-decoration:underline;}#yiv2925365401 a:visited, #yiv2925365401 span.yiv2925365401MsoHyperlinkFollowed {color:purple;text-decoration:underline;}#yiv2925365401 p.yiv2925365401msonormal0, #yiv2925365401 li.yiv2925365401msonormal0, #yiv2925365401 div.yiv2925365401msonormal0 {margin-right:0cm;margin-left:0cm;font-size:11.0pt;font-family:sans-serif;}#yiv2925365401 p.yiv2925365401msonormal, #yiv2925365401 li.yiv2925365401msonormal, #yiv2925365401 div.yiv2925365401msonormal {margin-right:0cm;margin-left:0cm;font-size:11.0pt;font-family:sans-serif;}#yiv2925365401 p.yiv2925365401msochpdefault, #yiv2925365401 li.yiv2925365401msochpdefault, #yiv2925365401 div.yiv2925365401msochpdefault {margin-right:0cm;margin-left:0cm;font-size:11.0pt;font-family:sans-serif;}#yiv2925365401 span.yiv2925365401msohyperlink {}#yiv2925365401 span.yiv2925365401msohyperlinkfollowed {}#yiv2925365401 span.yiv2925365401emailstyle17 {}#yiv2925365401 p.yiv2925365401msonormal1, #yiv2925365401 li.yiv2925365401msonormal1, #yiv2925365401 div.yiv2925365401msonormal1 {margin:0cm;margin-bottom:.0001pt;font-size:12.0pt;font-family:sans-serif;}#yiv2925365401 p.yiv2925365401msonormal2, #yiv2925365401 li.yiv2925365401msonormal2, #yiv2925365401 div.yiv2925365401msonormal2 {margin:0cm;margin-bottom:.0001pt;font-size:12.0pt;font-family:sans-serif;}#yiv2925365401 span.yiv2925365401msohyperlink1 {color:#0563C1;text-decoration:underline;}#yiv2925365401 span.yiv2925365401msohyperlinkfollowed1 {color:#954F72;text-decoration:underline;}#yiv2925365401 span.yiv2925365401emailstyle171 {font-family:sans-serif;color:windowtext;}#yiv2925365401 p.yiv2925365401msochpdefault1, #yiv2925365401 li.yiv2925365401msochpdefault1, #yiv2925365401 div.yiv2925365401msochpdefault1 {margin-right:0cm;margin-left:0cm;font-size:11.0pt;font-family:sans-serif;}#yiv2925365401 span.yiv2925365401EmailStyle29 {font-family:sans-serif;color:windowtext;}#yiv2925365401 .yiv2925365401MsoChpDefault {font-size:10.0pt;} _filtered #yiv2925365401 {margin:70.85pt 70.85pt 70.85pt 70.85pt;}#yiv2925365401 div.yiv2925365401WordSection1 {}#yiv2925365401 Hi Pasan,   Thanks for the reply but yes of course I have mi_fifo module load and configure.   I’m using also dispatcher with a db provisioning and without the db parameters for the rtpengine module I can manage the dispatcher entry via opensipsctl fi ds_XXXX with the the db parameters for the rtpengine module, command opensipsctl fifo ds_list provide the same error about mi_fifo.   thanks   De : Users au nom de Pasan Meemaduma via Users Répondre à : Pasan Meemaduma , OpenSIPS users mailling list Date : vendredi 6 juillet 2018 à 12:23 À : OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] RTPengine provisioning by db   Hi Alain,   do you have following config lines in your opensips.cfg file which enables the fifo management interface ?   loadmodule "mi_fifo.so" # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") According to the error you got it doesn't seems to exists.     On Friday, 6 July 2018, 3:27:51 PM GMT+5:30, Alain Bieuzent wrote:     Hi all,   I’m trying to provionning rtpengine via mysql db, but i can’t add a new proxy via opensipctl   My parameters are :   ###################################################################### #### RTPENGINE module ###################################################################### #modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 udp:10.207.201.20:2223") i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") modparam("rtpengine", "db_url", "mysql://opensips:XXXXXXXX at ZZZZZZZZZZZ/opensips_lb_did_2_4") modparam("rtpengine", "db_table", "rtpproxy_sockets") modparam("rtpengine", "socket_column", "rtpproxy_sock") modparam("rtpengine", "set_column", "set_id")   when i run : opensipsctl fifo rtpengine_show i received this error : ERROR: /tmp/opensips_fifo does not exist ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")' in your config ERROR: and also have loaded the mi_fifo module.   When i use « rtpengine_sock », it works   version: opensips 2.4.1 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. git revision: 5d042cffc main.c compiled on 09:17:20 Jul  6 2018 with gcc 4.7   Someone can help me ?   thanks _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From aqsyounas at gmail.com Fri Jul 6 14:21:44 2018 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 6 Jul 2018 23:21:44 +0500 Subject: [OpenSIPS-Users] Opensips not setting public ip when using mid_registrar module Message-ID: Greeting list, I have opensips behind the nat and using mid_registrar module with below configuration. listen=udp:10.111.222.4:5060 as 25.96.10.146:5060 loadmodule "mid_registrar.so" modparam("mid_registrar", "mode", 0) modparam("mid_registrar", "default_expires", 120) modparam("mid_registrar", "max_expires", 120) modparam("mid_registrar", "outgoing_expires", 7200) modparam("mid_registrar", "received_avp", "$avp(received)") And my Registration block. if (is_method("REGISTER")) { $du = "sip:41.76.34.144:5060"; mid_registrar_save("location"); switch ($retcode) { case 1: route(RELAY); break; default: xlog("L_INFO", "Failed"); } exit; } But I see Register leaving the server like below. 2018/07/06 18:11:59.343873 10.111.222.4:5060 -> 41.76.34.144:5060 REGISTER sip:captiveplanning.3cx.us:5060 SIP/2.0 Via: SIP/2.0/UDP 23.96.10.146:5060;branch=z9hG4bK7466.7191fac5.0 Via: SIP/2.0/UDP 10.0.0.125;rport=23144;received=x`;branch=z9hG4bKa981b751692BBD0A From: "Debra Gaglioti" ;tag=952308D0-E63A880D To: CSeq: 333 REGISTER Call-ID: da22e9a02781dcf350e2b1a5a313e358 *Contact: ;expires=120* User-Agent: PolycomVVX-VVX_301-UA/5.6.0.17325 Accept-Language: en-us,en;q=0.9 Proxy-Authorization: Digest username="devcmys", realm="3CXPhoneSystem", nonce="414d535c115041d173:6dd675ef072ae25029323dd3d0e0bb11", uri="sip: captiveplanning.3cx.us:5060", response="56c88b4 99d9e7da6dca69cfebdb23e", algorithm=MD5 Max-Forwards: 69 Expires: 120 Content-Length: 0 You can see contact header contains the IP of opensips listening interface not the public ip as advertised. Am i missing something or it is intended behavior? Any suggestion is much appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Mon Jul 9 00:34:30 2018 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Mon, 09 Jul 2018 01:34:30 -0300 Subject: [OpenSIPS-Users] mid_registrar work arround Message-ID: <1531110870.3207.5@smtp.networklab.ca> Hello Everyone, I have work around on this issue # Work arround for github issue #1109 append_hf("Path: \r\n" Right now we use 3 node cluster with 3 vips in active/active mode I need some idea how to determine path header listen ip address from 3 vips. volga629 From volga629 at networklab.ca Mon Jul 9 13:49:07 2018 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Mon, 09 Jul 2018 14:49:07 -0300 Subject: [OpenSIPS-Users] soket error Message-ID: <1531158547.3207.7@smtp.networklab.ca> Hello Everyone, Trying build test cluster active/active with 3 vips for each node separate virtual ip with keepalived And opensips failing to bound vip ip for corresponding vm This is interface config with vip 4: ens37: mtu 1500 qdisc fq state UNKNOWN group default qlen 1000 link/ether 00:50:56:a2:d5:0c brd ff:ff:ff:ff:ff:ff inet 10.100.104.4/28 brd 10.100.104.15 scope global ens37 valid_lft forever preferred_lft forever inet 10.100.104.7/28 scope global secondary ens37 ---> Keepalived VIP valid_lft forever preferred_lft forever inet6 fe80::b279:d4d6:45a1:6dd6/64 scope link valid_lft forever preferred_lft forever inet6 fe80::f9a7:9cc4:6a27:4cde/64 scope link dadfailed tentative valid_lft forever preferred_lft forever Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: Forwarding REGISTER to main registrar ~> [] Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:core:get_out_socket: no socket found Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:mid_registrar:uri2sock: no corresponding socket for af 2 Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:mid_registrar:overwrite_req_contacts: failed to obtain next hop socket, ci=0_3830065743 at 192.168.1.13 Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:mid_registrar:mid_reg_req_fwded: failed to overwrite Contact URIs Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:core:get_out_socket: no socket found Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 1 (no corresponding listening socket) Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:tm:t_forward_nonack: failure to add branches Listen listen=udp:10.100.104.7:5060 vip1 listen=udp:10.100.104.8:5060 vip2 listen=udp:10.100.104.9:5060 vip3 listen=bin:10.100.104.7:5585 bin cluster Any idea what what will be possible cause ? volga629 From pasandev at ymail.com Tue Jul 10 05:44:49 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Tue, 10 Jul 2018 09:44:49 +0000 (UTC) Subject: [OpenSIPS-Users] soket error In-Reply-To: <1531158547.3207.7@smtp.networklab.ca> References: <1531158547.3207.7@smtp.networklab.ca> Message-ID: <1882020822.1561348.1531215889633@mail.yahoo.com> Hi Volga, Its a very common question about non local address binding in linux. did you set following via sysctl ?net.ipv4.ip_nonlocal_bind=1 On Monday, 9 July 2018, 11:20:04 PM GMT+5:30, volga629 at networklab.ca wrote: Hello Everyone, Trying build test cluster active/active  with 3 vips for each node separate virtual ip with keepalived And opensips failing to bound vip ip for corresponding vm This is interface config with vip 4: ens37: mtu 1500 qdisc fq state UNKNOWN group default qlen 1000     link/ether 00:50:56:a2:d5:0c brd ff:ff:ff:ff:ff:ff     inet 10.100.104.4/28 brd 10.100.104.15 scope global ens37       valid_lft forever preferred_lft forever     inet 10.100.104.7/28 scope global secondary ens37                ---> Keepalived VIP       valid_lft forever preferred_lft forever     inet6 fe80::b279:d4d6:45a1:6dd6/64 scope link       valid_lft forever preferred_lft forever     inet6 fe80::f9a7:9cc4:6a27:4cde/64 scope link dadfailed tentative       valid_lft forever preferred_lft forever Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: Forwarding REGISTER to main registrar ~> [] Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:core:get_out_socket: no socket found Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:mid_registrar:uri2sock: no corresponding socket for af 2 Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:mid_registrar:overwrite_req_contacts: failed to obtain next hop socket, ci=0_3830065743 at 192.168.1.13 Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:mid_registrar:mid_reg_req_fwded: failed to overwrite Contact URIs Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:core:get_out_socket: no socket found Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 1  (no corresponding listening socket) Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: ERROR:tm:t_forward_nonack: failure to add branches Listen listen=udp:10.100.104.7:5060 vip1 listen=udp:10.100.104.8:5060 vip2 listen=udp:10.100.104.9:5060 vip3 listen=bin:10.100.104.7:5585 bin cluster Any idea what what will be possible cause ? volga629 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Tue Jul 10 07:20:54 2018 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Tue, 10 Jul 2018 08:20:54 -0300 Subject: [OpenSIPS-Users] soket error In-Reply-To: <1882020822.1561348.1531215889633@mail.yahoo.com> References: <1531158547.3207.7@smtp.networklab.ca> <1882020822.1561348.1531215889633@mail.yahoo.com> Message-ID: <1531221654.19705.0@smtp.networklab.ca> Hello Pasan, The whole issue is mhomed=1 it can't determine correct socket or don't have logic to work with vip addresses. I have to use on LAN side force_send_socket(udp:vip1:5060); But issue right now if one node goes down and vip is relocated to another node, how to determine which socket to use to send call from correct soure ip ? volga629 On Tue, Jul 10, 2018 at 6:44 AM, Pasan Meemaduma via Users wrote: > Hi Volga, > > Its a very common question about non local address binding in linux. > did you set following via sysctl ? > net.ipv4.ip_nonlocal_bind=1 > > > > On Monday, 9 July 2018, 11:20:04 PM GMT+5:30, volga629 at networklab.ca > wrote: > > > Hello Everyone, > Trying build test cluster active/active with 3 vips for each node > separate virtual ip with keepalived > And opensips failing to bound vip ip for corresponding vm > This is interface config with vip > > 4: ens37: mtu 1500 qdisc fq state > UNKNOWN group default qlen 1000 > link/ether 00:50:56:a2:d5:0c brd ff:ff:ff:ff:ff:ff > inet 10.100.104.4/28 brd 10.100.104.15 scope global ens37 > valid_lft forever preferred_lft forever > inet 10.100.104.7/28 scope global secondary ens37 > ---> Keepalived VIP > valid_lft forever preferred_lft forever > inet6 fe80::b279:d4d6:45a1:6dd6/64 scope link > valid_lft forever preferred_lft forever > inet6 fe80::f9a7:9cc4:6a27:4cde/64 scope link dadfailed tentative > valid_lft forever preferred_lft forever > > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: Forwarding REGISTER to > main registrar ~> [] > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:core:get_out_socket: no socket found > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:mid_registrar:uri2sock: no corresponding socket for af 2 > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:mid_registrar:overwrite_req_contacts: failed to obtain next hop > socket, ci=0_3830065743 at 192.168.1.13 > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:mid_registrar:mid_reg_req_fwded: failed to overwrite Contact > URIs > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:core:get_out_socket: no socket found > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 1 (no > corresponding listening socket) > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:tm:t_forward_nonack: failure to add branches > > > Listen > > listen=udp:10.100.104.7:5060 vip1 > listen=udp:10.100.104.8:5060 vip2 > listen=udp:10.100.104.9:5060 vip3 > listen=bin:10.100.104.7:5585 bin cluster > > Any idea what what will be possible cause ? > > volga629 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From pasandev at ymail.com Tue Jul 10 10:17:06 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Tue, 10 Jul 2018 14:17:06 +0000 (UTC) Subject: [OpenSIPS-Users] soket error In-Reply-To: <1531221654.19705.0@smtp.networklab.ca> References: <1531158547.3207.7@smtp.networklab.ca> <1882020822.1561348.1531215889633@mail.yahoo.com> <1531221654.19705.0@smtp.networklab.ca> Message-ID: <218785706.1664377.1531232227011@mail.yahoo.com> HI Volga, I haven't used mhomed=1 param, Apologies not sure how to help there. On Tuesday, 10 July 2018, 4:51:12 PM GMT+5:30, wrote: Hello Pasan, The whole issue is mhomed=1 it can't determine correct socket or don't have logic to work with vip addresses. I have to use on LAN side force_send_socket(udp:vip1:5060); But issue right now if one node goes down and vip is relocated to another node, how to determine which socket to use to send call from correct soure ip ? volga629 On Tue, Jul 10, 2018 at 6:44 AM, Pasan Meemaduma via Users wrote: > Hi Volga, > > Its a very common question about non local address binding in linux. > did you set following via sysctl ? > net.ipv4.ip_nonlocal_bind=1 > > > > On Monday, 9 July 2018, 11:20:04 PM GMT+5:30, volga629 at networklab.ca > wrote: > > > Hello Everyone, > Trying build test cluster active/active  with 3 vips for each node > separate virtual ip with keepalived > And opensips failing to bound vip ip for corresponding vm > This is interface config with vip > > 4: ens37: mtu 1500 qdisc fq state > UNKNOWN group default qlen 1000 >    link/ether 00:50:56:a2:d5:0c brd ff:ff:ff:ff:ff:ff >    inet 10.100.104.4/28 brd 10.100.104.15 scope global ens37 >      valid_lft forever preferred_lft forever >    inet 10.100.104.7/28 scope global secondary ens37 > ---> Keepalived VIP >      valid_lft forever preferred_lft forever >    inet6 fe80::b279:d4d6:45a1:6dd6/64 scope link >      valid_lft forever preferred_lft forever >    inet6 fe80::f9a7:9cc4:6a27:4cde/64 scope link dadfailed tentative >      valid_lft forever preferred_lft forever > > > Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: Forwarding REGISTER to > main registrar ~> [] > Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:core:get_out_socket: no socket found > Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:mid_registrar:uri2sock: no corresponding socket for af 2 > Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:mid_registrar:overwrite_req_contacts: failed to obtain next hop > socket, ci=0_3830065743 at 192.168.1.13 > Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:mid_registrar:mid_reg_req_fwded: failed to overwrite Contact > URIs > Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:core:get_out_socket: no socket found > Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 1  (no > corresponding listening socket) > Jul  9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > ERROR:tm:t_forward_nonack: failure to add branches > > > Listen > > listen=udp:10.100.104.7:5060 vip1 > listen=udp:10.100.104.8:5060 vip2 > listen=udp:10.100.104.9:5060 vip3 > listen=bin:10.100.104.7:5585 bin cluster > > Any idea what what will be possible cause ? > > volga629 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Wed Jul 11 08:55:00 2018 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Wed, 11 Jul 2018 14:55:00 +0200 Subject: [OpenSIPS-Users] RTPengine provisioning by db In-Reply-To: References: <342869488.15527.1530872281196@mail.yahoo.com> Message-ID: <1B1B1E03-192C-4E67-8D8F-A5FBBE119D37@free.fr> Hi all, nobody has any other idea? Thanks De : Users au nom de Alain Bieuzent Répondre à : OpenSIPS users mailling list Date : vendredi 6 juillet 2018 à 14:06 À : Pasan Meemaduma , OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] RTPengine provisioning by db Hi Pasan, Thanks for the reply but yes of course I have mi_fifo module load and configure. I’m using also dispatcher with a db provisioning and without the db parameters for the rtpengine module I can manage the dispatcher entry via opensipsctl fi ds_XXXX with the the db parameters for the rtpengine module, command opensipsctl fifo ds_list provide the same error about mi_fifo. thanks De : Users au nom de Pasan Meemaduma via Users Répondre à : Pasan Meemaduma , OpenSIPS users mailling list Date : vendredi 6 juillet 2018 à 12:23 À : OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] RTPengine provisioning by db Hi Alain, do you have following config lines in your opensips.cfg file which enables the fifo management interface ? loadmodule "mi_fifo.so" # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") According to the error you got it doesn't seems to exists. On Friday, 6 July 2018, 3:27:51 PM GMT+5:30, Alain Bieuzent wrote: Hi all, I’m trying to provionning rtpengine via mysql db, but i can’t add a new proxy via opensipctl My parameters are : ###################################################################### #### RTPENGINE module ###################################################################### #modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 udp:10.207.201.20:2223") i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") modparam("rtpengine", "db_url", "mysql://opensips:XXXXXXXX at ZZZZZZZZZZZ/opensips_lb_did_2_4") modparam("rtpengine", "db_table", "rtpproxy_sockets") modparam("rtpengine", "socket_column", "rtpproxy_sock") modparam("rtpengine", "set_column", "set_id") when i run : opensipsctl fifo rtpengine_show i received this error : ERROR: /tmp/opensips_fifo does not exist ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")' in your config ERROR: and also have loaded the mi_fifo module. When i use « rtpengine_sock », it works version: opensips 2.4.1 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. git revision: 5d042cffc main.c compiled on 09:17:20 Jul 6 2018 with gcc 4.7 Someone can help me ? thanks _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Jul 11 10:19:50 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 11 Jul 2018 17:19:50 +0300 Subject: [OpenSIPS-Users] RTPengine provisioning by db In-Reply-To: <1B1B1E03-192C-4E67-8D8F-A5FBBE119D37@free.fr> References: <342869488.15527.1530872281196@mail.yahoo.com> <1B1B1E03-192C-4E67-8D8F-A5FBBE119D37@free.fr> Message-ID: Hi, Alain! I think Pasan is right, you are either using a different /tmp/opensips_fifo modparam, or you can't access that file with the user you are trying to run. Or, perhaps you have a syntax error in your configuration file, and OpenSIPS doesn't start at all, that's why opensipsctl reports it cannot find the fifo file. If I am looking better at the snippet you posted, I see an "i" before a comment there: i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") That could be a parsing error which might prevent opensips from starting, thus creating the fifo file. Did you check if the file is actually there after you get the opensipsctl error? Best regards, Răzvan On 07/11/2018 03:55 PM, Alain Bieuzent wrote: > Hi all, > > nobody has any other idea? > > Thanks > > *De : *Users au nom de Alain Bieuzent > > *Répondre à : *OpenSIPS users mailling list > *Date : *vendredi 6 juillet 2018 à 14:06 > *À : *Pasan Meemaduma , OpenSIPS users mailling list > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > Hi Pasan, > > Thanks for the reply but yes of course I have mi_fifo module load and > configure. > > I’m using also dispatcher with a db provisioning and without the db > parameters for the rtpengine module I can manage the dispatcher entry > via opensipsctl fi ds_XXXX > > with the the db parameters for the rtpengine module, command opensipsctl > fifo ds_list provide the same error about mi_fifo. > > thanks > > *De : *Users au nom de Pasan > Meemaduma via Users > *Répondre à : *Pasan Meemaduma , OpenSIPS users > mailling list > *Date : *vendredi 6 juillet 2018 à 12:23 > *À : *OpenSIPS users mailling list > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > Hi Alain, > > do you have following config lines in your opensips.cfg file which > enables the fifo management interface ? > > loadmodule "mi_fifo.so" > > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") > > According to the error you got it doesn't seems to exists. > > On Friday, 6 July 2018, 3:27:51 PM GMT+5:30, Alain Bieuzent > wrote: > > Hi all, > > I’m trying to provionning rtpengine via mysql db, but i can’t add a new > proxy via opensipctl > > My parameters are : > > ###################################################################### > > #### RTPENGINE module > > ###################################################################### > > #modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 > udp:10.207.201.20:2223") > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > modparam("rtpengine", "db_url", > "mysql://opensips:XXXXXXXX at ZZZZZZZZZZZ/opensips_lb_did_2_4") > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > modparam("rtpengine", "set_column", "set_id") > > when i run : opensipsctl fifo rtpengine_show > > i received this error : > > ERROR: /tmp/opensips_fifo does not exist > > ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", > "/tmp/opensips_fifo")' in your config > > ERROR: and also have loaded the mi_fifo module. > > When i use « rtpengine_sock », it works > > version: opensips 2.4.1 (x86_64/linux) > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll, sigio_rt, select. > > git revision: 5d042cffc > > main.c compiled on 09:17:20 Jul  6 2018 with gcc 4.7 > > Someone can help me ? > > thanks > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From alain.bieuzent at free.fr Wed Jul 11 11:37:21 2018 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Wed, 11 Jul 2018 17:37:21 +0200 Subject: [OpenSIPS-Users] RTPengine provisioning by db In-Reply-To: References: <342869488.15527.1530872281196@mail.yahoo.com> <1B1B1E03-192C-4E67-8D8F-A5FBBE119D37@free.fr> Message-ID: <99E76416-9DE1-478F-9268-91143F414366@free.fr> Hi, Rasvan! I double check and the typo with the "i" is not the cause. I used these parameters: ###################################################################### #### RTPENGINE module ###################################################################### modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 udp:10.207.201.20:2223") modparam("rtpengine", "db_url", "mysql://opensips:XXXX at ZZZ/opensips_lb_did_2_4") modparam("rtpengine", "db_table", "rtpproxy_sockets") modparam("rtpengine", "socket_column", "rtpproxy_sock") modparam("rtpengine", "set_column", "set_id") If i comment the 4 last line to do not use db, opensips start i can use opensipsctl fifo ds_list (so no issue with fifo file) If i comment only the first line to use db, opensipsctl produce the error about fifo file. I don't change anything else except comment the first or the last 4 line. I don't know where to look .. thanks for your help Le 11/07/2018 16:24, « Users au nom de Răzvan Crainea » a écrit : Hi, Alain! I think Pasan is right, you are either using a different /tmp/opensips_fifo modparam, or you can't access that file with the user you are trying to run. Or, perhaps you have a syntax error in your configuration file, and OpenSIPS doesn't start at all, that's why opensipsctl reports it cannot find the fifo file. If I am looking better at the snippet you posted, I see an "i" before a comment there: i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") That could be a parsing error which might prevent opensips from starting, thus creating the fifo file. Did you check if the file is actually there after you get the opensipsctl error? Best regards, Răzvan On 07/11/2018 03:55 PM, Alain Bieuzent wrote: > Hi all, > > nobody has any other idea? > > Thanks > > *De : *Users au nom de Alain Bieuzent > > *Répondre à : *OpenSIPS users mailling list > *Date : *vendredi 6 juillet 2018 à 14:06 > *À : *Pasan Meemaduma , OpenSIPS users mailling list > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > Hi Pasan, > > Thanks for the reply but yes of course I have mi_fifo module load and > configure. > > I’m using also dispatcher with a db provisioning and without the db > parameters for the rtpengine module I can manage the dispatcher entry > via opensipsctl fi ds_XXXX > > with the the db parameters for the rtpengine module, command opensipsctl > fifo ds_list provide the same error about mi_fifo. > > thanks > > *De : *Users au nom de Pasan > Meemaduma via Users > *Répondre à : *Pasan Meemaduma , OpenSIPS users > mailling list > *Date : *vendredi 6 juillet 2018 à 12:23 > *À : *OpenSIPS users mailling list > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > Hi Alain, > > do you have following config lines in your opensips.cfg file which > enables the fifo management interface ? > > loadmodule "mi_fifo.so" > > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") > > According to the error you got it doesn't seems to exists. > > On Friday, 6 July 2018, 3:27:51 PM GMT+5:30, Alain Bieuzent > wrote: > > Hi all, > > I’m trying to provionning rtpengine via mysql db, but i can’t add a new > proxy via opensipctl > > My parameters are : > > ###################################################################### > > #### RTPENGINE module > > ###################################################################### > > #modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 > udp:10.207.201.20:2223") > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > modparam("rtpengine", "db_url", > "mysql://opensips:XXXXXXXX at ZZZZZZZZZZZ/opensips_lb_did_2_4") > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > modparam("rtpengine", "set_column", "set_id") > > when i run : opensipsctl fifo rtpengine_show > > i received this error : > > ERROR: /tmp/opensips_fifo does not exist > > ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", > "/tmp/opensips_fifo")' in your config > > ERROR: and also have loaded the mi_fifo module. > > When i use « rtpengine_sock », it works > > version: opensips 2.4.1 (x86_64/linux) > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll, sigio_rt, select. > > git revision: 5d042cffc > > main.c compiled on 09:17:20 Jul 6 2018 with gcc 4.7 > > Someone can help me ? > > thanks > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Wed Jul 11 11:43:35 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 11 Jul 2018 18:43:35 +0300 Subject: [OpenSIPS-Users] RTPengine provisioning by db In-Reply-To: <99E76416-9DE1-478F-9268-91143F414366@free.fr> References: <342869488.15527.1530872281196@mail.yahoo.com> <1B1B1E03-192C-4E67-8D8F-A5FBBE119D37@free.fr> <99E76416-9DE1-478F-9268-91143F414366@free.fr> Message-ID: Hello! After you comment the first line, can you make sure opensips actually starts (ps aux | grep opensips). After that, can you check that the /tmp/opensips_fifo file exists? Best regards, Răzvan On 07/11/2018 06:37 PM, Alain Bieuzent wrote: > Hi, Rasvan! > > > > I double check and the typo with the "i" is not the cause. > > > > I used these parameters: > > ###################################################################### > > #### RTPENGINE module > > ###################################################################### > > modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 udp:10.207.201.20:2223") > > > > modparam("rtpengine", "db_url", "mysql://opensips:XXXX at ZZZ/opensips_lb_did_2_4") > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > modparam("rtpengine", "set_column", "set_id") > > > > If i comment the 4 last line to do not use db, opensips start i can use opensipsctl fifo ds_list (so no issue with fifo file) > > If i comment only the first line to use db, opensipsctl produce the error about fifo file. > > I don't change anything else except comment the first or the last 4 line. > > > > I don't know where to look .. > > > > thanks for your help > > > > > > Le 11/07/2018 16:24, « Users au nom de Răzvan Crainea » a écrit : > > > > Hi, Alain! > > > > I think Pasan is right, you are either using a different > > /tmp/opensips_fifo modparam, or you can't access that file with the user > > you are trying to run. Or, perhaps you have a syntax error in your > > configuration file, and OpenSIPS doesn't start at all, that's why > > opensipsctl reports it cannot find the fifo file. > > If I am looking better at the snippet you posted, I see an "i" before a > > comment there: > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > > > That could be a parsing error which might prevent opensips from > > starting, thus creating the fifo file. > > > > Did you check if the file is actually there after you get the > > opensipsctl error? > > > > Best regards, > > Răzvan > > > > On 07/11/2018 03:55 PM, Alain Bieuzent wrote: > > > Hi all, > > > > > > nobody has any other idea? > > > > > > Thanks > > > > > > *De : *Users au nom de Alain Bieuzent > > > > > > *Répondre à : *OpenSIPS users mailling list > > > *Date : *vendredi 6 juillet 2018 à 14:06 > > > *À : *Pasan Meemaduma , OpenSIPS users mailling list > > > > > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > > > > > Hi Pasan, > > > > > > Thanks for the reply but yes of course I have mi_fifo module load and > > > configure. > > > > > > I’m using also dispatcher with a db provisioning and without the db > > > parameters for the rtpengine module I can manage the dispatcher entry > > > via opensipsctl fi ds_XXXX > > > > > > with the the db parameters for the rtpengine module, command opensipsctl > > > fifo ds_list provide the same error about mi_fifo. > > > > > > thanks > > > > > > *De : *Users au nom de Pasan > > > Meemaduma via Users > > > *Répondre à : *Pasan Meemaduma , OpenSIPS users > > > mailling list > > > *Date : *vendredi 6 juillet 2018 à 12:23 > > > *À : *OpenSIPS users mailling list > > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > > > > > Hi Alain, > > > > > > do you have following config lines in your opensips.cfg file which > > > enables the fifo management interface ? > > > > > > loadmodule "mi_fifo.so" > > > > > > > > > # ----- mi_fifo params ----- > > > modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") > > > > > > According to the error you got it doesn't seems to exists. > > > > > > On Friday, 6 July 2018, 3:27:51 PM GMT+5:30, Alain Bieuzent > > > wrote: > > > > > > Hi all, > > > > > > I’m trying to provionning rtpengine via mysql db, but i can’t add a new > > > proxy via opensipctl > > > > > > My parameters are : > > > > > > ###################################################################### > > > > > > #### RTPENGINE module > > > > > > ###################################################################### > > > > > > #modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 > > > udp:10.207.201.20:2223") > > > > > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > > > > > modparam("rtpengine", "db_url", > > > "mysql://opensips:XXXXXXXX at ZZZZZZZZZZZ/opensips_lb_did_2_4") > > > > > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > > > > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > > > > > modparam("rtpengine", "set_column", "set_id") > > > > > > when i run : opensipsctl fifo rtpengine_show > > > > > > i received this error : > > > > > > ERROR: /tmp/opensips_fifo does not exist > > > > > > ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", > > > "/tmp/opensips_fifo")' in your config > > > > > > ERROR: and also have loaded the mi_fifo module. > > > > > > When i use « rtpengine_sock », it works > > > > > > version: opensips 2.4.1 (x86_64/linux) > > > > > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > > > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > > > > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > > > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > > > > > poll method support: poll, epoll, sigio_rt, select. > > > > > > git revision: 5d042cffc > > > > > > main.c compiled on 09:17:20 Jul 6 2018 with gcc 4.7 > > > > > > Someone can help me ? > > > > > > thanks > > > > > > _______________________________________________ > > > Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > _______________________________________________ > > > Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Răzvan Crainea > > OpenSIPS Core Developer > > http://www.opensips-solutions.com > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From alain.bieuzent at free.fr Wed Jul 11 11:59:45 2018 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Wed, 11 Jul 2018 17:59:45 +0200 Subject: [OpenSIPS-Users] RTPengine provisioning by db In-Reply-To: References: <342869488.15527.1530872281196@mail.yahoo.com> <1B1B1E03-192C-4E67-8D8F-A5FBBE119D37@free.fr> <99E76416-9DE1-478F-9268-91143F414366@free.fr> Message-ID: <137C7E29-F73E-4E54-BF1B-B8F0892274DD@free.fr> never forgot to check the logs..... yes, you are right opensips didn't start with this error: Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:db_check_table_version: invalid version 0 for table rtpproxy_sockets found, expected 1 Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:rtpengine:mod_init: error during table version check Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:init_mod: failed to initialize module rtpengine Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:main: error while initializing modules My fault sorry for the noise. Thanks, Razvan. Le 11/07/2018 17:48, « Users au nom de Răzvan Crainea » a écrit : Hello! After you comment the first line, can you make sure opensips actually starts (ps aux | grep opensips). After that, can you check that the /tmp/opensips_fifo file exists? Best regards, Răzvan On 07/11/2018 06:37 PM, Alain Bieuzent wrote: > Hi, Rasvan! > > > > I double check and the typo with the "i" is not the cause. > > > > I used these parameters: > > ###################################################################### > > #### RTPENGINE module > > ###################################################################### > > modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 udp:10.207.201.20:2223") > > > > modparam("rtpengine", "db_url", "mysql://opensips:XXXX at ZZZ/opensips_lb_did_2_4") > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > modparam("rtpengine", "set_column", "set_id") > > > > If i comment the 4 last line to do not use db, opensips start i can use opensipsctl fifo ds_list (so no issue with fifo file) > > If i comment only the first line to use db, opensipsctl produce the error about fifo file. > > I don't change anything else except comment the first or the last 4 line. > > > > I don't know where to look .. > > > > thanks for your help > > > > > > Le 11/07/2018 16:24, « Users au nom de Răzvan Crainea » a écrit : > > > > Hi, Alain! > > > > I think Pasan is right, you are either using a different > > /tmp/opensips_fifo modparam, or you can't access that file with the user > > you are trying to run. Or, perhaps you have a syntax error in your > > configuration file, and OpenSIPS doesn't start at all, that's why > > opensipsctl reports it cannot find the fifo file. > > If I am looking better at the snippet you posted, I see an "i" before a > > comment there: > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > > > That could be a parsing error which might prevent opensips from > > starting, thus creating the fifo file. > > > > Did you check if the file is actually there after you get the > > opensipsctl error? > > > > Best regards, > > Răzvan > > > > On 07/11/2018 03:55 PM, Alain Bieuzent wrote: > > > Hi all, > > > > > > nobody has any other idea? > > > > > > Thanks > > > > > > *De : *Users au nom de Alain Bieuzent > > > > > > *Répondre à : *OpenSIPS users mailling list > > > *Date : *vendredi 6 juillet 2018 à 14:06 > > > *À : *Pasan Meemaduma , OpenSIPS users mailling list > > > > > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > > > > > Hi Pasan, > > > > > > Thanks for the reply but yes of course I have mi_fifo module load and > > > configure. > > > > > > I’m using also dispatcher with a db provisioning and without the db > > > parameters for the rtpengine module I can manage the dispatcher entry > > > via opensipsctl fi ds_XXXX > > > > > > with the the db parameters for the rtpengine module, command opensipsctl > > > fifo ds_list provide the same error about mi_fifo. > > > > > > thanks > > > > > > *De : *Users au nom de Pasan > > > Meemaduma via Users > > > *Répondre à : *Pasan Meemaduma , OpenSIPS users > > > mailling list > > > *Date : *vendredi 6 juillet 2018 à 12:23 > > > *À : *OpenSIPS users mailling list > > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > > > > > Hi Alain, > > > > > > do you have following config lines in your opensips.cfg file which > > > enables the fifo management interface ? > > > > > > loadmodule "mi_fifo.so" > > > > > > > > > # ----- mi_fifo params ----- > > > modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") > > > > > > According to the error you got it doesn't seems to exists. > > > > > > On Friday, 6 July 2018, 3:27:51 PM GMT+5:30, Alain Bieuzent > > > wrote: > > > > > > Hi all, > > > > > > I’m trying to provionning rtpengine via mysql db, but i can’t add a new > > > proxy via opensipctl > > > > > > My parameters are : > > > > > > ###################################################################### > > > > > > #### RTPENGINE module > > > > > > ###################################################################### > > > > > > #modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 > > > udp:10.207.201.20:2223") > > > > > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > > > > > modparam("rtpengine", "db_url", > > > "mysql://opensips:XXXXXXXX at ZZZZZZZZZZZ/opensips_lb_did_2_4") > > > > > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > > > > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > > > > > modparam("rtpengine", "set_column", "set_id") > > > > > > when i run : opensipsctl fifo rtpengine_show > > > > > > i received this error : > > > > > > ERROR: /tmp/opensips_fifo does not exist > > > > > > ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", > > > "/tmp/opensips_fifo")' in your config > > > > > > ERROR: and also have loaded the mi_fifo module. > > > > > > When i use « rtpengine_sock », it works > > > > > > version: opensips 2.4.1 (x86_64/linux) > > > > > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > > > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > > > > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > > > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > > > > > poll method support: poll, epoll, sigio_rt, select. > > > > > > git revision: 5d042cffc > > > > > > main.c compiled on 09:17:20 Jul 6 2018 with gcc 4.7 > > > > > > Someone can help me ? > > > > > > thanks > > > > > > _______________________________________________ > > > Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > _______________________________________________ > > > Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Răzvan Crainea > > OpenSIPS Core Developer > > http://www.opensips-solutions.com > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Wed Jul 11 11:57:49 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 11 Jul 2018 18:57:49 +0300 Subject: [OpenSIPS-Users] RTPengine provisioning by db In-Reply-To: <137C7E29-F73E-4E54-BF1B-B8F0892274DD@free.fr> References: <342869488.15527.1530872281196@mail.yahoo.com> <1B1B1E03-192C-4E67-8D8F-A5FBBE119D37@free.fr> <99E76416-9DE1-478F-9268-91143F414366@free.fr> <137C7E29-F73E-4E54-BF1B-B8F0892274DD@free.fr> Message-ID: <59ec43a8-3e33-877f-3cd4-39815a215611@opensips.org> No problem, thanks for sharing the initial cause! Best regards, Răzvan On 07/11/2018 06:59 PM, Alain Bieuzent wrote: > never forgot to check the logs..... > > yes, you are right opensips didn't start with this error: > > > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:db_check_table_version: invalid version 0 for table rtpproxy_sockets found, expected 1 > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:rtpengine:mod_init: error during table version check > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:init_mod: failed to initialize module rtpengine > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:main: error while initializing modules > > > > My fault sorry for the noise. > > > > Thanks, Razvan. > > > > Le 11/07/2018 17:48, « Users au nom de Răzvan Crainea » a écrit : > > > > Hello! > > > > After you comment the first line, can you make sure opensips actually > > starts (ps aux | grep opensips). After that, can you check that the > > /tmp/opensips_fifo file exists? > > > > Best regards, > > Răzvan > > > > On 07/11/2018 06:37 PM, Alain Bieuzent wrote: > > > Hi, Rasvan! > > > > > > > > > > > > I double check and the typo with the "i" is not the cause. > > > > > > > > > > > > I used these parameters: > > > > > > ###################################################################### > > > > > > #### RTPENGINE module > > > > > > ###################################################################### > > > > > > modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 udp:10.207.201.20:2223") > > > > > > > > > > > > modparam("rtpengine", "db_url", "mysql://opensips:XXXX at ZZZ/opensips_lb_did_2_4") > > > > > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > > > > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > > > > > modparam("rtpengine", "set_column", "set_id") > > > > > > > > > > > > If i comment the 4 last line to do not use db, opensips start i can use opensipsctl fifo ds_list (so no issue with fifo file) > > > > > > If i comment only the first line to use db, opensipsctl produce the error about fifo file. > > > > > > I don't change anything else except comment the first or the last 4 line. > > > > > > > > > > > > I don't know where to look .. > > > > > > > > > > > > thanks for your help > > > > > > > > > > > > > > > > > > Le 11/07/2018 16:24, « Users au nom de Răzvan Crainea » a écrit : > > > > > > > > > > > > Hi, Alain! > > > > > > > > > > > > I think Pasan is right, you are either using a different > > > > > > /tmp/opensips_fifo modparam, or you can't access that file with the user > > > > > > you are trying to run. Or, perhaps you have a syntax error in your > > > > > > configuration file, and OpenSIPS doesn't start at all, that's why > > > > > > opensipsctl reports it cannot find the fifo file. > > > > > > If I am looking better at the snippet you posted, I see an "i" before a > > > > > > comment there: > > > > > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > > > > > > > > > > > That could be a parsing error which might prevent opensips from > > > > > > starting, thus creating the fifo file. > > > > > > > > > > > > Did you check if the file is actually there after you get the > > > > > > opensipsctl error? > > > > > > > > > > > > Best regards, > > > > > > Răzvan > > > > > > > > > > > > On 07/11/2018 03:55 PM, Alain Bieuzent wrote: > > > > > > > Hi all, > > > > > > > > > > > > > > nobody has any other idea? > > > > > > > > > > > > > > Thanks > > > > > > > > > > > > > > *De : *Users au nom de Alain Bieuzent > > > > > > > > > > > > > > *Répondre à : *OpenSIPS users mailling list > > > > > > > *Date : *vendredi 6 juillet 2018 à 14:06 > > > > > > > *À : *Pasan Meemaduma , OpenSIPS users mailling list > > > > > > > > > > > > > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > > > > > > > > > > > > > Hi Pasan, > > > > > > > > > > > > > > Thanks for the reply but yes of course I have mi_fifo module load and > > > > > > > configure. > > > > > > > > > > > > > > I’m using also dispatcher with a db provisioning and without the db > > > > > > > parameters for the rtpengine module I can manage the dispatcher entry > > > > > > > via opensipsctl fi ds_XXXX > > > > > > > > > > > > > > with the the db parameters for the rtpengine module, command opensipsctl > > > > > > > fifo ds_list provide the same error about mi_fifo. > > > > > > > > > > > > > > thanks > > > > > > > > > > > > > > *De : *Users au nom de Pasan > > > > > > > Meemaduma via Users > > > > > > > *Répondre à : *Pasan Meemaduma , OpenSIPS users > > > > > > > mailling list > > > > > > > *Date : *vendredi 6 juillet 2018 à 12:23 > > > > > > > *À : *OpenSIPS users mailling list > > > > > > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > > > > > > > > > > > > > Hi Alain, > > > > > > > > > > > > > > do you have following config lines in your opensips.cfg file which > > > > > > > enables the fifo management interface ? > > > > > > > > > > > > > > loadmodule "mi_fifo.so" > > > > > > > > > > > > > > > > > > > > > # ----- mi_fifo params ----- > > > > > > > modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") > > > > > > > > > > > > > > According to the error you got it doesn't seems to exists. > > > > > > > > > > > > > > On Friday, 6 July 2018, 3:27:51 PM GMT+5:30, Alain Bieuzent > > > > > > > wrote: > > > > > > > > > > > > > > Hi all, > > > > > > > > > > > > > > I’m trying to provionning rtpengine via mysql db, but i can’t add a new > > > > > > > proxy via opensipctl > > > > > > > > > > > > > > My parameters are : > > > > > > > > > > > > > > ###################################################################### > > > > > > > > > > > > > > #### RTPENGINE module > > > > > > > > > > > > > > ###################################################################### > > > > > > > > > > > > > > #modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 > > > > > > > udp:10.207.201.20:2223") > > > > > > > > > > > > > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > > > > > > > > > > > > > modparam("rtpengine", "db_url", > > > > > > > "mysql://opensips:XXXXXXXX at ZZZZZZZZZZZ/opensips_lb_did_2_4") > > > > > > > > > > > > > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > > > > > > > > > > > > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > > > > > > > > > > > > > modparam("rtpengine", "set_column", "set_id") > > > > > > > > > > > > > > when i run : opensipsctl fifo rtpengine_show > > > > > > > > > > > > > > i received this error : > > > > > > > > > > > > > > ERROR: /tmp/opensips_fifo does not exist > > > > > > > > > > > > > > ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", > > > > > > > "/tmp/opensips_fifo")' in your config > > > > > > > > > > > > > > ERROR: and also have loaded the mi_fifo module. > > > > > > > > > > > > > > When i use « rtpengine_sock », it works > > > > > > > > > > > > > > version: opensips 2.4.1 (x86_64/linux) > > > > > > > > > > > > > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > > > > > > > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > > > > > > > > > > > > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > > > > > > > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > > > > > > > > > > > > > poll method support: poll, epoll, sigio_rt, select. > > > > > > > > > > > > > > git revision: 5d042cffc > > > > > > > > > > > > > > main.c compiled on 09:17:20 Jul 6 2018 with gcc 4.7 > > > > > > > > > > > > > > Someone can help me ? > > > > > > > > > > > > > > thanks > > > > > > > > > > > > > > _______________________________________________ > > > > > > > Users mailing list > > > > > > > Users at lists.opensips.org > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > _______________________________________________ Users mailing list > > > > > > > Users at lists.opensips.org > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > _______________________________________________ Users mailing list > > > > > > > Users at lists.opensips.org > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > Users mailing list > > > > > > > Users at lists.opensips.org > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > -- > > > > > > Răzvan Crainea > > > > > > OpenSIPS Core Developer > > > > > > http://www.opensips-solutions.com > > > > > > > > > > > > _______________________________________________ > > > > > > Users mailing list > > > > > > Users at lists.opensips.org > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Răzvan Crainea > > OpenSIPS Core Developer > > http://www.opensips-solutions.com > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From volga629 at networklab.ca Wed Jul 11 12:14:51 2018 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Wed, 11 Jul 2018 13:14:51 -0300 Subject: [OpenSIPS-Users] soket error In-Reply-To: <218785706.1664377.1531232227011@mail.yahoo.com> References: <1531158547.3207.7@smtp.networklab.ca> <1882020822.1561348.1531215889633@mail.yahoo.com> <1531221654.19705.0@smtp.networklab.ca> <218785706.1664377.1531232227011@mail.yahoo.com> Message-ID: <1531325691.19705.2@smtp.networklab.ca> Thank you for reply, on top dialog replication getting those messages where bin was set listen actual lan interface not vip. Jul 11 10:28:12 aitossbc01 /usr/sbin/opensips[5318]: WARNING:dialog:fetch_socket_info: non-local socket ...ignoring Jul 11 10:28:12 aitossbc01 /usr/sbin/opensips[5318]: ERROR:dialog:dlg_replicated_create: Replicated dialog doesn't match any listening sockets Jul 11 10:28:12 aitossbc01 /usr/sbin/opensips[5318]: ERROR:dialog:receive_dlg_repl: Failed to process a binary packet! volga629 On Tue, Jul 10, 2018 at 11:17 AM, Pasan Meemaduma wrote: > HI Volga, > > I haven't used mhomed=1 param, Apologies not sure how to help there. > > > On Tuesday, 10 July 2018, 4:51:12 PM GMT+5:30, > wrote: > > > Hello Pasan, > The whole issue is mhomed=1 it can't determine correct socket or don't > have logic to work with vip addresses. I have to use on LAN side > > force_send_socket(udp:vip1:5060); > > But issue right now if one node goes down and vip is relocated to > another node, how to determine which socket to use to send call from > correct soure ip ? > > > volga629 > > On Tue, Jul 10, 2018 at 6:44 AM, Pasan Meemaduma via Users > wrote: > > Hi Volga, > > > > Its a very common question about non local address binding in linux. > > did you set following via sysctl ? > > net.ipv4.ip_nonlocal_bind=1 > > > > > > > > On Monday, 9 July 2018, 11:20:04 PM GMT+5:30, volga629 at networklab.ca > > wrote: > > > > > > Hello Everyone, > > Trying build test cluster active/active with 3 vips for each node > > separate virtual ip with keepalived > > And opensips failing to bound vip ip for corresponding vm > > This is interface config with vip > > > > 4: ens37: mtu 1500 qdisc fq state > > UNKNOWN group default qlen 1000 > > link/ether 00:50:56:a2:d5:0c brd ff:ff:ff:ff:ff:ff > > inet 10.100.104.4/28 brd 10.100.104.15 scope global ens37 > > valid_lft forever preferred_lft forever > > inet 10.100.104.7/28 scope global secondary ens37 > > ---> Keepalived VIP > > valid_lft forever preferred_lft forever > > inet6 fe80::b279:d4d6:45a1:6dd6/64 scope link > > valid_lft forever preferred_lft forever > > inet6 fe80::f9a7:9cc4:6a27:4cde/64 scope link dadfailed tentative > > valid_lft forever preferred_lft forever > > > > > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: Forwarding REGISTER > to > > main registrar ~> [] > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:core:get_out_socket: no socket found > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:mid_registrar:uri2sock: no corresponding socket for af 2 > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:mid_registrar:overwrite_req_contacts: failed to obtain next > hop > > socket, ci=0_3830065743 at 192.168.1.13 > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:mid_registrar:mid_reg_req_fwded: failed to overwrite Contact > > URIs > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:core:get_out_socket: no socket found > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 1 (no > > corresponding listening socket) > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:tm:t_forward_nonack: failure to add branches > > > > > > Listen > > > > listen=udp:10.100.104.7:5060 vip1 > > listen=udp:10.100.104.8:5060 vip2 > > listen=udp:10.100.104.9:5060 vip3 > > listen=bin:10.100.104.7:5585 bin cluster > > > > Any idea what what will be possible cause ? > > > > volga629 > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From alain.bieuzent at free.fr Thu Jul 12 05:56:57 2018 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Thu, 12 Jul 2018 11:56:57 +0200 Subject: [OpenSIPS-Users] RTPengine provisioning by db In-Reply-To: <59ec43a8-3e33-877f-3cd4-39815a215611@opensips.org> References: <342869488.15527.1530872281196@mail.yahoo.com> <1B1B1E03-192C-4E67-8D8F-A5FBBE119D37@free.fr> <99E76416-9DE1-478F-9268-91143F414366@free.fr> <137C7E29-F73E-4E54-BF1B-B8F0892274DD@free.fr> <59ec43a8-3e33-877f-3cd4-39815a215611@opensips.org> Message-ID: Hi all, I just created a new database and version of table rtpproxy_sockets is 0, so i think there is a mismatch somewhere because opensips 2.4.1 expected 1. Is there somewhere the mysql schema of 2.4 version? thanks Le 11/07/2018 18:02, « Users au nom de Răzvan Crainea » a écrit : No problem, thanks for sharing the initial cause! Best regards, Răzvan On 07/11/2018 06:59 PM, Alain Bieuzent wrote: > never forgot to check the logs..... > > yes, you are right opensips didn't start with this error: > > > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:db_check_table_version: invalid version 0 for table rtpproxy_sockets found, expected 1 > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:rtpengine:mod_init: error during table version check > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:init_mod: failed to initialize module rtpengine > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:main: error while initializing modules > > > > My fault sorry for the noise. > > > > Thanks, Razvan. > > > > Le 11/07/2018 17:48, « Users au nom de Răzvan Crainea » a écrit : > > > > Hello! > > > > After you comment the first line, can you make sure opensips actually > > starts (ps aux | grep opensips). After that, can you check that the > > /tmp/opensips_fifo file exists? > > > > Best regards, > > Răzvan > > > > On 07/11/2018 06:37 PM, Alain Bieuzent wrote: > > > Hi, Rasvan! > > > > > > > > > > > > I double check and the typo with the "i" is not the cause. > > > > > > > > > > > > I used these parameters: > > > > > > ###################################################################### > > > > > > #### RTPENGINE module > > > > > > ###################################################################### > > > > > > modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 udp:10.207.201.20:2223") > > > > > > > > > > > > modparam("rtpengine", "db_url", "mysql://opensips:XXXX at ZZZ/opensips_lb_did_2_4") > > > > > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > > > > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > > > > > modparam("rtpengine", "set_column", "set_id") > > > > > > > > > > > > If i comment the 4 last line to do not use db, opensips start i can use opensipsctl fifo ds_list (so no issue with fifo file) > > > > > > If i comment only the first line to use db, opensipsctl produce the error about fifo file. > > > > > > I don't change anything else except comment the first or the last 4 line. > > > > > > > > > > > > I don't know where to look .. > > > > > > > > > > > > thanks for your help > > > > > > > > > > > > > > > > > > Le 11/07/2018 16:24, « Users au nom de Răzvan Crainea » a écrit : > > > > > > > > > > > > Hi, Alain! > > > > > > > > > > > > I think Pasan is right, you are either using a different > > > > > > /tmp/opensips_fifo modparam, or you can't access that file with the user > > > > > > you are trying to run. Or, perhaps you have a syntax error in your > > > > > > configuration file, and OpenSIPS doesn't start at all, that's why > > > > > > opensipsctl reports it cannot find the fifo file. > > > > > > If I am looking better at the snippet you posted, I see an "i" before a > > > > > > comment there: > > > > > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > > > > > > > > > > > That could be a parsing error which might prevent opensips from > > > > > > starting, thus creating the fifo file. > > > > > > > > > > > > Did you check if the file is actually there after you get the > > > > > > opensipsctl error? > > > > > > > > > > > > Best regards, > > > > > > Răzvan > > > > > > > > > > > > On 07/11/2018 03:55 PM, Alain Bieuzent wrote: > > > > > > > Hi all, > > > > > > > > > > > > > > nobody has any other idea? > > > > > > > > > > > > > > Thanks > > > > > > > > > > > > > > *De : *Users au nom de Alain Bieuzent > > > > > > > > > > > > > > *Répondre à : *OpenSIPS users mailling list > > > > > > > *Date : *vendredi 6 juillet 2018 à 14:06 > > > > > > > *À : *Pasan Meemaduma , OpenSIPS users mailling list > > > > > > > > > > > > > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > > > > > > > > > > > > > Hi Pasan, > > > > > > > > > > > > > > Thanks for the reply but yes of course I have mi_fifo module load and > > > > > > > configure. > > > > > > > > > > > > > > I’m using also dispatcher with a db provisioning and without the db > > > > > > > parameters for the rtpengine module I can manage the dispatcher entry > > > > > > > via opensipsctl fi ds_XXXX > > > > > > > > > > > > > > with the the db parameters for the rtpengine module, command opensipsctl > > > > > > > fifo ds_list provide the same error about mi_fifo. > > > > > > > > > > > > > > thanks > > > > > > > > > > > > > > *De : *Users au nom de Pasan > > > > > > > Meemaduma via Users > > > > > > > *Répondre à : *Pasan Meemaduma , OpenSIPS users > > > > > > > mailling list > > > > > > > *Date : *vendredi 6 juillet 2018 à 12:23 > > > > > > > *À : *OpenSIPS users mailling list > > > > > > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > > > > > > > > > > > > > Hi Alain, > > > > > > > > > > > > > > do you have following config lines in your opensips.cfg file which > > > > > > > enables the fifo management interface ? > > > > > > > > > > > > > > loadmodule "mi_fifo.so" > > > > > > > > > > > > > > > > > > > > > # ----- mi_fifo params ----- > > > > > > > modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") > > > > > > > > > > > > > > According to the error you got it doesn't seems to exists. > > > > > > > > > > > > > > On Friday, 6 July 2018, 3:27:51 PM GMT+5:30, Alain Bieuzent > > > > > > > wrote: > > > > > > > > > > > > > > Hi all, > > > > > > > > > > > > > > I’m trying to provionning rtpengine via mysql db, but i can’t add a new > > > > > > > proxy via opensipctl > > > > > > > > > > > > > > My parameters are : > > > > > > > > > > > > > > ###################################################################### > > > > > > > > > > > > > > #### RTPENGINE module > > > > > > > > > > > > > > ###################################################################### > > > > > > > > > > > > > > #modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 > > > > > > > udp:10.207.201.20:2223") > > > > > > > > > > > > > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > > > > > > > > > > > > > modparam("rtpengine", "db_url", > > > > > > > "mysql://opensips:XXXXXXXX at ZZZZZZZZZZZ/opensips_lb_did_2_4") > > > > > > > > > > > > > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > > > > > > > > > > > > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > > > > > > > > > > > > > modparam("rtpengine", "set_column", "set_id") > > > > > > > > > > > > > > when i run : opensipsctl fifo rtpengine_show > > > > > > > > > > > > > > i received this error : > > > > > > > > > > > > > > ERROR: /tmp/opensips_fifo does not exist > > > > > > > > > > > > > > ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", > > > > > > > "/tmp/opensips_fifo")' in your config > > > > > > > > > > > > > > ERROR: and also have loaded the mi_fifo module. > > > > > > > > > > > > > > When i use « rtpengine_sock », it works > > > > > > > > > > > > > > version: opensips 2.4.1 (x86_64/linux) > > > > > > > > > > > > > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > > > > > > > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > > > > > > > > > > > > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > > > > > > > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > > > > > > > > > > > > > poll method support: poll, epoll, sigio_rt, select. > > > > > > > > > > > > > > git revision: 5d042cffc > > > > > > > > > > > > > > main.c compiled on 09:17:20 Jul 6 2018 with gcc 4.7 > > > > > > > > > > > > > > Someone can help me ? > > > > > > > > > > > > > > thanks > > > > > > > > > > > > > > _______________________________________________ > > > > > > > Users mailing list > > > > > > > Users at lists.opensips.org > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > _______________________________________________ Users mailing list > > > > > > > Users at lists.opensips.org > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > _______________________________________________ Users mailing list > > > > > > > Users at lists.opensips.org > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > Users mailing list > > > > > > > Users at lists.opensips.org > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > -- > > > > > > Răzvan Crainea > > > > > > OpenSIPS Core Developer > > > > > > http://www.opensips-solutions.com > > > > > > > > > > > > _______________________________________________ > > > > > > Users mailing list > > > > > > Users at lists.opensips.org > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Răzvan Crainea > > OpenSIPS Core Developer > > http://www.opensips-solutions.com > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From alain.bieuzent at free.fr Thu Jul 12 08:30:10 2018 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Thu, 12 Jul 2018 14:30:10 +0200 Subject: [OpenSIPS-Users] RTPengine provisioning by db In-Reply-To: References: <342869488.15527.1530872281196@mail.yahoo.com> <1B1B1E03-192C-4E67-8D8F-A5FBBE119D37@free.fr> <99E76416-9DE1-478F-9268-91143F414366@free.fr> <137C7E29-F73E-4E54-BF1B-B8F0892274DD@free.fr> <59ec43a8-3e33-877f-3cd4-39815a215611@opensips.org> Message-ID: Hi Razvan, are you sure this new support of provisioning via DB is supported by tools "opensipctldb create", because new table "rtpengine" does not exist when I create a new database from scratch. thanks https://github.com/OpenSIPS/opensips/commit/094c850dd91286ff75bf3463dafbf83a398aaf8f Le 12/07/2018 11:58, « Users au nom de Alain Bieuzent » a écrit : Hi all, I just created a new database and version of table rtpproxy_sockets is 0, so i think there is a mismatch somewhere because opensips 2.4.1 expected 1. Is there somewhere the mysql schema of 2.4 version? thanks Le 11/07/2018 18:02, « Users au nom de Răzvan Crainea » a écrit : No problem, thanks for sharing the initial cause! Best regards, Răzvan On 07/11/2018 06:59 PM, Alain Bieuzent wrote: > never forgot to check the logs..... > > yes, you are right opensips didn't start with this error: > > > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:db_check_table_version: invalid version 0 for table rtpproxy_sockets found, expected 1 > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:rtpengine:mod_init: error during table version check > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:init_mod: failed to initialize module rtpengine > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:main: error while initializing modules > > > > My fault sorry for the noise. > > > > Thanks, Razvan. > > > > Le 11/07/2018 17:48, « Users au nom de Răzvan Crainea » a écrit : > > > > Hello! > > > > After you comment the first line, can you make sure opensips actually > > starts (ps aux | grep opensips). After that, can you check that the > > /tmp/opensips_fifo file exists? > > > > Best regards, > > Răzvan > > > > On 07/11/2018 06:37 PM, Alain Bieuzent wrote: > > > Hi, Rasvan! > > > > > > > > > > > > I double check and the typo with the "i" is not the cause. > > > > > > > > > > > > I used these parameters: > > > > > > ###################################################################### > > > > > > #### RTPENGINE module > > > > > > ###################################################################### > > > > > > modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 udp:10.207.201.20:2223") > > > > > > > > > > > > modparam("rtpengine", "db_url", "mysql://opensips:XXXX at ZZZ/opensips_lb_did_2_4") > > > > > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > > > > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > > > > > modparam("rtpengine", "set_column", "set_id") > > > > > > > > > > > > If i comment the 4 last line to do not use db, opensips start i can use opensipsctl fifo ds_list (so no issue with fifo file) > > > > > > If i comment only the first line to use db, opensipsctl produce the error about fifo file. > > > > > > I don't change anything else except comment the first or the last 4 line. > > > > > > > > > > > > I don't know where to look .. > > > > > > > > > > > > thanks for your help > > > > > > > > > > > > > > > > > > Le 11/07/2018 16:24, « Users au nom de Răzvan Crainea » a écrit : > > > > > > > > > > > > Hi, Alain! > > > > > > > > > > > > I think Pasan is right, you are either using a different > > > > > > /tmp/opensips_fifo modparam, or you can't access that file with the user > > > > > > you are trying to run. Or, perhaps you have a syntax error in your > > > > > > configuration file, and OpenSIPS doesn't start at all, that's why > > > > > > opensipsctl reports it cannot find the fifo file. > > > > > > If I am looking better at the snippet you posted, I see an "i" before a > > > > > > comment there: > > > > > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > > > > > > > > > > > That could be a parsing error which might prevent opensips from > > > > > > starting, thus creating the fifo file. > > > > > > > > > > > > Did you check if the file is actually there after you get the > > > > > > opensipsctl error? > > > > > > > > > > > > Best regards, > > > > > > Răzvan > > > > > > > > > > > > On 07/11/2018 03:55 PM, Alain Bieuzent wrote: > > > > > > > Hi all, > > > > > > > > > > > > > > nobody has any other idea? > > > > > > > > > > > > > > Thanks > > > > > > > > > > > > > > *De : *Users au nom de Alain Bieuzent > > > > > > > > > > > > > > *Répondre à : *OpenSIPS users mailling list > > > > > > > *Date : *vendredi 6 juillet 2018 à 14:06 > > > > > > > *À : *Pasan Meemaduma , OpenSIPS users mailling list > > > > > > > > > > > > > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > > > > > > > > > > > > > Hi Pasan, > > > > > > > > > > > > > > Thanks for the reply but yes of course I have mi_fifo module load and > > > > > > > configure. > > > > > > > > > > > > > > I’m using also dispatcher with a db provisioning and without the db > > > > > > > parameters for the rtpengine module I can manage the dispatcher entry > > > > > > > via opensipsctl fi ds_XXXX > > > > > > > > > > > > > > with the the db parameters for the rtpengine module, command opensipsctl > > > > > > > fifo ds_list provide the same error about mi_fifo. > > > > > > > > > > > > > > thanks > > > > > > > > > > > > > > *De : *Users au nom de Pasan > > > > > > > Meemaduma via Users > > > > > > > *Répondre à : *Pasan Meemaduma , OpenSIPS users > > > > > > > mailling list > > > > > > > *Date : *vendredi 6 juillet 2018 à 12:23 > > > > > > > *À : *OpenSIPS users mailling list > > > > > > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > > > > > > > > > > > > > Hi Alain, > > > > > > > > > > > > > > do you have following config lines in your opensips.cfg file which > > > > > > > enables the fifo management interface ? > > > > > > > > > > > > > > loadmodule "mi_fifo.so" > > > > > > > > > > > > > > > > > > > > > # ----- mi_fifo params ----- > > > > > > > modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") > > > > > > > > > > > > > > According to the error you got it doesn't seems to exists. > > > > > > > > > > > > > > On Friday, 6 July 2018, 3:27:51 PM GMT+5:30, Alain Bieuzent > > > > > > > wrote: > > > > > > > > > > > > > > Hi all, > > > > > > > > > > > > > > I’m trying to provionning rtpengine via mysql db, but i can’t add a new > > > > > > > proxy via opensipctl > > > > > > > > > > > > > > My parameters are : > > > > > > > > > > > > > > ###################################################################### > > > > > > > > > > > > > > #### RTPENGINE module > > > > > > > > > > > > > > ###################################################################### > > > > > > > > > > > > > > #modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 > > > > > > > udp:10.207.201.20:2223") > > > > > > > > > > > > > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > > > > > > > > > > > > > modparam("rtpengine", "db_url", > > > > > > > "mysql://opensips:XXXXXXXX at ZZZZZZZZZZZ/opensips_lb_did_2_4") > > > > > > > > > > > > > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > > > > > > > > > > > > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > > > > > > > > > > > > > modparam("rtpengine", "set_column", "set_id") > > > > > > > > > > > > > > when i run : opensipsctl fifo rtpengine_show > > > > > > > > > > > > > > i received this error : > > > > > > > > > > > > > > ERROR: /tmp/opensips_fifo does not exist > > > > > > > > > > > > > > ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", > > > > > > > "/tmp/opensips_fifo")' in your config > > > > > > > > > > > > > > ERROR: and also have loaded the mi_fifo module. > > > > > > > > > > > > > > When i use « rtpengine_sock », it works > > > > > > > > > > > > > > version: opensips 2.4.1 (x86_64/linux) > > > > > > > > > > > > > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > > > > > > > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > > > > > > > > > > > > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > > > > > > > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > > > > > > > > > > > > > poll method support: poll, epoll, sigio_rt, select. > > > > > > > > > > > > > > git revision: 5d042cffc > > > > > > > > > > > > > > main.c compiled on 09:17:20 Jul 6 2018 with gcc 4.7 > > > > > > > > > > > > > > Someone can help me ? > > > > > > > > > > > > > > thanks > > > > > > > > > > > > > > _______________________________________________ > > > > > > > Users mailing list > > > > > > > Users at lists.opensips.org > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > _______________________________________________ Users mailing list > > > > > > > Users at lists.opensips.org > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > _______________________________________________ Users mailing list > > > > > > > Users at lists.opensips.org > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > Users mailing list > > > > > > > Users at lists.opensips.org > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > -- > > > > > > Răzvan Crainea > > > > > > OpenSIPS Core Developer > > > > > > http://www.opensips-solutions.com > > > > > > > > > > > > _______________________________________________ > > > > > > Users mailing list > > > > > > Users at lists.opensips.org > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Răzvan Crainea > > OpenSIPS Core Developer > > http://www.opensips-solutions.com > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Thu Jul 12 08:30:54 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 12 Jul 2018 15:30:54 +0300 Subject: [OpenSIPS-Users] RTPengine provisioning by db In-Reply-To: References: <342869488.15527.1530872281196@mail.yahoo.com> <1B1B1E03-192C-4E67-8D8F-A5FBBE119D37@free.fr> <99E76416-9DE1-478F-9268-91143F414366@free.fr> <137C7E29-F73E-4E54-BF1B-B8F0892274DD@free.fr> <59ec43a8-3e33-877f-3cd4-39815a215611@opensips.org> Message-ID: <5268043a-67f3-bfe3-b86a-7608a1623c56@opensips.org> Hi, Alain! No, rtpengine is not created when you run `opensipsctldb create`. You'll have to create it yourself using something like: mysql -uUSER -pPASS -dDB -hHOST < scripts/mysql/rtpengine-create.sql Best regards, Răzvan On 07/12/2018 03:30 PM, Alain Bieuzent wrote: > Hi Razvan, > > > > are you sure this new support of provisioning via DB is supported by tools "opensipctldb create", because new table "rtpengine" does not exist when I create a new database from scratch. > > > > thanks > > > > https://github.com/OpenSIPS/opensips/commit/094c850dd91286ff75bf3463dafbf83a398aaf8f > > > > > > > > > > Le 12/07/2018 11:58, « Users au nom de Alain Bieuzent » a écrit : > > > > Hi all, > > > > > > > > I just created a new database and version of table rtpproxy_sockets is 0, so i think there is a mismatch somewhere because opensips 2.4.1 expected 1. > > > > > > > > Is there somewhere the mysql schema of 2.4 version? > > > > > > > > thanks > > > > > > > > Le 11/07/2018 18:02, « Users au nom de Răzvan Crainea » a écrit : > > > > > > > > No problem, thanks for sharing the initial cause! > > > > > > > > Best regards, > > > > Răzvan > > > > > > > > On 07/11/2018 06:59 PM, Alain Bieuzent wrote: > > > > > never forgot to check the logs..... > > > > > > > > > > yes, you are right opensips didn't start with this error: > > > > > > > > > > > > > > > > > > > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:db_check_table_version: invalid version 0 for table rtpproxy_sockets found, expected 1 > > > > > > > > > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:rtpengine:mod_init: error during table version check > > > > > > > > > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:init_mod: failed to initialize module rtpengine > > > > > > > > > > Jul 11 17:52:21 opensips-test opensips[20625]: ERROR:core:main: error while initializing modules > > > > > > > > > > > > > > > > > > > > My fault sorry for the noise. > > > > > > > > > > > > > > > > > > > > Thanks, Razvan. > > > > > > > > > > > > > > > > > > > > Le 11/07/2018 17:48, « Users au nom de Răzvan Crainea » a écrit : > > > > > > > > > > > > > > > > > > > > Hello! > > > > > > > > > > > > > > > > > > > > After you comment the first line, can you make sure opensips actually > > > > > > > > > > starts (ps aux | grep opensips). After that, can you check that the > > > > > > > > > > /tmp/opensips_fifo file exists? > > > > > > > > > > > > > > > > > > > > Best regards, > > > > > > > > > > Răzvan > > > > > > > > > > > > > > > > > > > > On 07/11/2018 06:37 PM, Alain Bieuzent wrote: > > > > > > > > > > > Hi, Rasvan! > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > I double check and the typo with the "i" is not the cause. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > I used these parameters: > > > > > > > > > > > > > > > > > > > > > > ###################################################################### > > > > > > > > > > > > > > > > > > > > > > #### RTPENGINE module > > > > > > > > > > > > > > > > > > > > > > ###################################################################### > > > > > > > > > > > > > > > > > > > > > > modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 udp:10.207.201.20:2223") > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > modparam("rtpengine", "db_url", "mysql://opensips:XXXX at ZZZ/opensips_lb_did_2_4") > > > > > > > > > > > > > > > > > > > > > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > > > > > > > > > > > > > > > > > > > > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > > > > > > > > > > > > > > > > > > > > > modparam("rtpengine", "set_column", "set_id") > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > If i comment the 4 last line to do not use db, opensips start i can use opensipsctl fifo ds_list (so no issue with fifo file) > > > > > > > > > > > > > > > > > > > > > > If i comment only the first line to use db, opensipsctl produce the error about fifo file. > > > > > > > > > > > > > > > > > > > > > > I don't change anything else except comment the first or the last 4 line. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > I don't know where to look .. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > thanks for your help > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Le 11/07/2018 16:24, « Users au nom de Răzvan Crainea » a écrit : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Hi, Alain! > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > I think Pasan is right, you are either using a different > > > > > > > > > > > > > > > > > > > > > > /tmp/opensips_fifo modparam, or you can't access that file with the user > > > > > > > > > > > > > > > > > > > > > > you are trying to run. Or, perhaps you have a syntax error in your > > > > > > > > > > > > > > > > > > > > > > configuration file, and OpenSIPS doesn't start at all, that's why > > > > > > > > > > > > > > > > > > > > > > opensipsctl reports it cannot find the fifo file. > > > > > > > > > > > > > > > > > > > > > > If I am looking better at the snippet you posted, I see an "i" before a > > > > > > > > > > > > > > > > > > > > > > comment there: > > > > > > > > > > > > > > > > > > > > > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > That could be a parsing error which might prevent opensips from > > > > > > > > > > > > > > > > > > > > > > starting, thus creating the fifo file. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Did you check if the file is actually there after you get the > > > > > > > > > > > > > > > > > > > > > > opensipsctl error? > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Best regards, > > > > > > > > > > > > > > > > > > > > > > Răzvan > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On 07/11/2018 03:55 PM, Alain Bieuzent wrote: > > > > > > > > > > > > > > > > > > > > > > > Hi all, > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > nobody has any other idea? > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Thanks > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > *De : *Users au nom de Alain Bieuzent > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > *Répondre à : *OpenSIPS users mailling list > > > > > > > > > > > > > > > > > > > > > > > *Date : *vendredi 6 juillet 2018 à 14:06 > > > > > > > > > > > > > > > > > > > > > > > *À : *Pasan Meemaduma , OpenSIPS users mailling list > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Hi Pasan, > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Thanks for the reply but yes of course I have mi_fifo module load and > > > > > > > > > > > > > > > > > > > > > > > configure. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > I’m using also dispatcher with a db provisioning and without the db > > > > > > > > > > > > > > > > > > > > > > > parameters for the rtpengine module I can manage the dispatcher entry > > > > > > > > > > > > > > > > > > > > > > > via opensipsctl fi ds_XXXX > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > with the the db parameters for the rtpengine module, command opensipsctl > > > > > > > > > > > > > > > > > > > > > > > fifo ds_list provide the same error about mi_fifo. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > thanks > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > *De : *Users au nom de Pasan > > > > > > > > > > > > > > > > > > > > > > > Meemaduma via Users > > > > > > > > > > > > > > > > > > > > > > > *Répondre à : *Pasan Meemaduma , OpenSIPS users > > > > > > > > > > > > > > > > > > > > > > > mailling list > > > > > > > > > > > > > > > > > > > > > > > *Date : *vendredi 6 juillet 2018 à 12:23 > > > > > > > > > > > > > > > > > > > > > > > *À : *OpenSIPS users mailling list > > > > > > > > > > > > > > > > > > > > > > > *Objet : *Re: [OpenSIPS-Users] RTPengine provisioning by db > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Hi Alain, > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > do you have following config lines in your opensips.cfg file which > > > > > > > > > > > > > > > > > > > > > > > enables the fifo management interface ? > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > loadmodule "mi_fifo.so" > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > # ----- mi_fifo params ----- > > > > > > > > > > > > > > > > > > > > > > > modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > According to the error you got it doesn't seems to exists. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Friday, 6 July 2018, 3:27:51 PM GMT+5:30, Alain Bieuzent > > > > > > > > > > > > > > > > > > > > > > > wrote: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Hi all, > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > I’m trying to provionning rtpengine via mysql db, but i can’t add a new > > > > > > > > > > > > > > > > > > > > > > > proxy via opensipctl > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > My parameters are : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ###################################################################### > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > #### RTPENGINE module > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ###################################################################### > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > #modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.19:2223 > > > > > > > > > > > > > > > > > > > > > > > udp:10.207.201.20:2223") > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > i#modparam("rtpengine", "rtpengine_sock", "udp:10.207.201.20:2223") > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > modparam("rtpengine", "db_url", > > > > > > > > > > > > > > > > > > > > > > > "mysql://opensips:XXXXXXXX at ZZZZZZZZZZZ/opensips_lb_did_2_4") > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > modparam("rtpengine", "db_table", "rtpproxy_sockets") > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > modparam("rtpengine", "socket_column", "rtpproxy_sock") > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > modparam("rtpengine", "set_column", "set_id") > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > when i run : opensipsctl fifo rtpengine_show > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > i received this error : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ERROR: /tmp/opensips_fifo does not exist > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", > > > > > > > > > > > > > > > > > > > > > > > "/tmp/opensips_fifo")' in your config > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ERROR: and also have loaded the mi_fifo module. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > When i use « rtpengine_sock », it works > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > version: opensips 2.4.1 (x86_64/linux) > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > > > > > > > > > > > > > > > > > > > > > > > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > > > > > > > > > > > > > > > > > > > > > > > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > poll method support: poll, epoll, sigio_rt, select. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > git revision: 5d042cffc > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > main.c compiled on 09:17:20 Jul 6 2018 with gcc 4.7 > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Someone can help me ? > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > thanks > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > > > > > > > > > > > > > > > > > Users mailing list > > > > > > > > > > > > > > > > > > > > > > > Users at lists.opensips.org > > > > > > > > > > > > > > > > > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ Users mailing list > > > > > > > > > > > > > > > > > > > > > > > Users at lists.opensips.org > > > > > > > > > > > > > > > > > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ Users mailing list > > > > > > > > > > > > > > > > > > > > > > > Users at lists.opensips.org > > > > > > > > > > > > > > > > > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > > > > > > > > > > > > > > > > > Users mailing list > > > > > > > > > > > > > > > > > > > > > > > Users at lists.opensips.org > > > > > > > > > > > > > > > > > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > > > > > > > > > > > > > > > > > > > Răzvan Crainea > > > > > > > > > > > > > > > > > > > > > > OpenSIPS Core Developer > > > > > > > > > > > > > > > > > > > > > > http://www.opensips-solutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > > > > > > > > > > > > > > > > Users mailing list > > > > > > > > > > > > > > > > > > > > > > Users at lists.opensips.org > > > > > > > > > > > > > > > > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > > > > > Users mailing list > > > > > > > > > > > Users at lists.opensips.org > > > > > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > > > > > > > Răzvan Crainea > > > > > > > > > > OpenSIPS Core Developer > > > > > > > > > > http://www.opensips-solutions.com > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > > > > Users mailing list > > > > > > > > > > Users at lists.opensips.org > > > > > > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > Users mailing list > > > > > Users at lists.opensips.org > > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > -- > > > > Răzvan Crainea > > > > OpenSIPS Core Developer > > > > http://www.opensips-solutions.com > > > > > > > > _______________________________________________ > > > > Users mailing list > > > > Users at lists.opensips.org > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From yuval.dinari at vonage.com Thu Jul 12 09:07:19 2018 From: yuval.dinari at vonage.com (Yuval Dinari) Date: Thu, 12 Jul 2018 16:07:19 +0300 Subject: [OpenSIPS-Users] Opensips TCP children deadlock Message-ID: Hi, I have a state in which opensips gets into an unrecoverable bad state, in which some of the tcp children process are stuck waiting to acquire a lock which they never get. The issue occurs in the following load test scenario: 1. About 25K clients register in TCP (but also happens with less) 2. All the TCP connections become unresponsive (by blocking outgoing traffic on the test clients machine) 3. INVITEs are sent for each of those clients, putting their connection in retransmit mode 4. After a few minutes opensips gets into a bad state - some tcp children run at 90-100% cpu, no traffic is being sent from the machine (including OPTIONS pings) 5. After all the tcp connections die due to timeouts, opensips does not recover, the mentioned symptoms stay 6. After all the registered users are removed from internal table there's still no change When attaching debugger to the problematic processes (with high cpu usage) we see that they're all stuck trying to get a lock which they never seem to get. Stack traces: #0 0x00007fd6b72d1bb7 in sched_yield () at ../sysdeps/unix/syscall-template.S:81 #1 0x0000000000549e65 in get_lock (lock=) at net/proto_tcp/../../net/../fastlock.h:221 #2 _tcp_write_on_socket (len=, buf=, fd=, c=) at net/proto_tcp/proto_tcp.c:724 #3 proto_tcp_send (send_sock=0x7ffd8e12c140, buf=0x0, len=399, to=0x7fd5c7ccdcc0, id=1) at net/proto_tcp/proto_tcp.c:922 #4 0x00007fd5a5cb7b30 in msg_send (msg=, len=, buf=, id=, to=, proto=, send_sock=0x7fd6a7208168) at ../../forward.h:123 #5 send_pr_buffer (rb=0x7fd5c7ccdca0, buf=0x7fd6a76b4a50, len=0, ctx=0xffffffffffffffff) at t_funcs.c:66 And: #0 0x00007fd6b72d1bb7 in sched_yield () at ../sysdeps/unix/syscall-template.S:81 #1 0x00000000005349b8 in get_lock (lock=) at net/../fastlock.h:221 #2 handle_io (event_type=, idx=, fm=) at net/net_tcp_proc.c:210 #3 io_wait_loop_epoll (repeat=287, t=, h=) at net/../io_wait_loop.h:280 This traces look the same every time we attach. The machine opensips runs on has 4 cpus. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From tito at xsvoce.com Thu Jul 12 13:27:27 2018 From: tito at xsvoce.com (Tito Cumpen) Date: Thu, 12 Jul 2018 10:27:27 -0700 Subject: [OpenSIPS-Users] Docker centos Message-ID: Hello, I've made an image of opensips in centos 7 but one of my concerns is that docker uses internal IP addresses. With this being the case is there any way to make opensips use stun to figure out its public IP? I tried referencing the GitHub docker-opensips which runs in Debian but I cannot find anything in there addresses this issue. Thanks, Tito -------------- next part -------------- An HTML attachment was scrubbed... URL: From ksrigo at gmail.com Fri Jul 13 03:52:37 2018 From: ksrigo at gmail.com (KSrigo) Date: Fri, 13 Jul 2018 09:52:37 +0200 Subject: [OpenSIPS-Users] Billing stack with Opensips and Cgrates Message-ID: <185F0446-2926-42B4-97D4-0899E1E432FC@gmail.com> Hi, Recently I have been working on implementing a Billing system at iBrowse. Throughout the process I wrote a tutorial. For those who are interested, I’m sharing it here: https://github.com/ksrigo/billing_system Please let me know your feedback. Voice Engineer Srigo Kana -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Fri Jul 13 05:11:16 2018 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 13 Jul 2018 12:11:16 +0300 Subject: [OpenSIPS-Users] Billing stack with Opensips and Cgrates In-Reply-To: <185F0446-2926-42B4-97D4-0899E1E432FC@gmail.com> References: <185F0446-2926-42B4-97D4-0899E1E432FC@gmail.com> Message-ID: <8f02f61e-3479-debd-24b4-767c1ed2cfd9@opensips.org> This is awesome, Srigo!  I'd be curious to see if there are any throughput differences between your "syslog - Filebeat - Logstash" stack vs. a "db_flatstore - " approach.  Nevertheless, this is the proper way to do it: de-coupled from SIP and de-coupled from DB. PS: I am the proud owner of your project's 1st Star Cheers, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 13.07.2018 10:52, KSrigo wrote: > Hi, > > Recently I have been working on implementing a Billing system at > iBrowse. Throughout the process I wrote a tutorial. > For those who are interested, I’m sharing it here: > > https://github.com/ksrigo/billing_system > > Please let me know your feedback. > > /Voice Engineer/ > *Srigo Kana* > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From sastre.sebastian at gmail.com Fri Jul 13 14:19:15 2018 From: sastre.sebastian at gmail.com (Sebastian Sastre) Date: Fri, 13 Jul 2018 14:19:15 -0400 Subject: [OpenSIPS-Users] Proto WSS No Open TCP Connection after reinvite Message-ID: Hello, I’ve been experiencing a situation with Proto WSS. The scenario is very simple. A call is established from an Asterisk Box to Opensips (UDP) and finally a SipJs7.8 (WSS). Everything works great and we are able to register using mid registrar and pass calls thru. When an agent puts the call on hold a reinvite is correctly negotiated and the call is placed on hold and viceversa. However!, if the originating caller disconnects the call while still on hold, Asterisk will correctly terminate the dialog with a Bye but when OpenSIPs will complain about not finding a suitable tcp connection and responds with a 477 even after successfully matching and processing the dialog termination correctly. opensipsctl fifo list_tcp_conns shows the connection available. The only way I found of fixing this problem is by adding fix_route_dialog() on the sequential loose route. if (loose_route()) { if (is_method("BYE")) { if (!validate_dialog()){ fix_route_dialog(); } What do you guys think? Am I messing up something in the script or is this the correct way to address this problem? The funny thing is that there is no difference notable between the bye after hold and a regular bye without putting the call on hold. Here is the opensips log with the error and the trace. https://pastebin.com/BEJ6fAR8 Thanks ! -------------- next part -------------- An HTML attachment was scrubbed... URL: From ksrigo at gmail.com Sat Jul 14 06:27:04 2018 From: ksrigo at gmail.com (Srigo Kanapathipillai) Date: Sat, 14 Jul 2018 12:27:04 +0200 Subject: [OpenSIPS-Users] Billing stack with Opensips and Cgrates In-Reply-To: <8f02f61e-3479-debd-24b4-767c1ed2cfd9@opensips.org> References: <185F0446-2926-42B4-97D4-0899E1E432FC@gmail.com> <8f02f61e-3479-debd-24b4-767c1ed2cfd9@opensips.org> Message-ID: <30236A0C-6207-4BC2-BF3F-080BBD110B09@gmail.com> Thank you Liviu!! I’m happy you like it, Its motivate me to write my projects. To answer your question, Dan asked me almost the same question: https://github.com/ksrigo/billing_system/issues/1 . I think db_flatestore + CDRC is much simpler at least from maintaining perspective (we have less services to maintain). As soon as I get some details from Dan, I will test it and let you know. Voice Engineer Srigo Kana > On 13 Jul 2018, at 11:11, Liviu Chircu wrote: > > This is awesome, Srigo! I'd be curious to see if there are any throughput differences between your "syslog - Filebeat - Logstash" stack vs. a "db_flatstore - " approach. Nevertheless, this is the proper way to do it: de-coupled from SIP and de-coupled from DB. > > PS: I am the proud owner of your project's 1st Star > > Cheers, > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > On 13.07.2018 10:52, KSrigo wrote: >> Hi, >> >> Recently I have been working on implementing a Billing system at iBrowse. Throughout the process I wrote a tutorial. >> For those who are interested, I’m sharing it here: >> >> https://github.com/ksrigo/billing_system >> >> Please let me know your feedback. >> >> Voice Engineer >> Srigo Kana >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From basit.engg at gmail.com Sat Jul 14 14:38:29 2018 From: basit.engg at gmail.com (Abdul Basit) Date: Sat, 14 Jul 2018 23:38:29 +0500 Subject: [OpenSIPS-Users] Billing stack with Opensips and Cgrates In-Reply-To: <30236A0C-6207-4BC2-BF3F-080BBD110B09@gmail.com> References: <185F0446-2926-42B4-97D4-0899E1E432FC@gmail.com> <8f02f61e-3479-debd-24b4-767c1ed2cfd9@opensips.org> <30236A0C-6207-4BC2-BF3F-080BBD110B09@gmail.com> Message-ID: Nice work and especially sharing of your work. Keep it up. -- regards, abdul basit On Sat, 14 Jul 2018 at 15:30, Srigo Kanapathipillai wrote: > Thank you Liviu!! I’m happy you like it, Its motivate me to write my > projects. > > To answer your question, Dan asked me almost the same question: > https://github.com/ksrigo/billing_system/issues/1. I think db_flatestore > + CDRC is much simpler at least from maintaining perspective (we have less > services to maintain). As soon as I get some details from Dan, I will test > it and let you know. > > *Voice Engineer* > *Srigo Kana* > > > On 13 Jul 2018, at 11:11, Liviu Chircu wrote: > > This is awesome, Srigo! I'd be curious to see if there are any throughput > differences between your "syslog - Filebeat - Logstash" stack vs. a > "db_flatstore - " approach. Nevertheless, > this is the proper way to do it: de-coupled from SIP and de-coupled from DB. > > PS: I am the proud owner of your project's 1st Star > > Cheers, > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 13.07.2018 10:52, KSrigo wrote: > > Hi, > > Recently I have been working on implementing a Billing system at iBrowse. > Throughout the process I wrote a tutorial. > For those who are interested, I’m sharing it here: > > https://github.com/ksrigo/billing_system > > Please let me know your feedback. > > *Voice Engineer* > *Srigo Kana* > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From basit.engg at gmail.com Sat Jul 14 21:12:02 2018 From: basit.engg at gmail.com (Abdul Basit) Date: Sun, 15 Jul 2018 06:12:02 +0500 Subject: [OpenSIPS-Users] check_blacklist function Question In-Reply-To: <62dbeaa8-5ccf-c26f-aea3-cadf855710f1@opensips.org> References: <79a34483-1bd6-71e4-1819-e89b6a7d3931@opensips.org> <62dbeaa8-5ccf-c26f-aea3-cadf855710f1@opensips.org> Message-ID: Hi Team, I have a scenario where I have to block list certain numbers (premium ones). I populated these numbers with in "globalblacklist" table. I followed documentation: http://www.opensips.org/html/docs/modules/2.4.x/userblacklist.html#func_check_blacklist While part of destination number defined as prefix, like 1437, blacklisting works and all the calls starting with destination number '1437' will block. id prefix whitelist description 1 1437 0 NULL But for the required case, if I add full number in prefix column, like 14375436291, opensips will not block the call. id prefix whitelist description 1 14375436291 0 NULL How can i block list of full numbers through opensips? -- regards, abdul basit -------------- next part -------------- An HTML attachment was scrubbed... URL: From basit.engg at gmail.com Sat Jul 14 23:18:44 2018 From: basit.engg at gmail.com (Abdul Basit) Date: Sun, 15 Jul 2018 08:18:44 +0500 Subject: [OpenSIPS-Users] check_blacklist function Question In-Reply-To: References: <79a34483-1bd6-71e4-1819-e89b6a7d3931@opensips.org> <62dbeaa8-5ccf-c26f-aea3-cadf855710f1@opensips.org> Message-ID: Never mind. Issue has been fixed. I was overlooking the number. Few digits were different in next request that fail the exact number match and causing call to go through. -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On Sun, 15 Jul 2018 at 06:12, Abdul Basit wrote: > Hi Team, > I have a scenario where I have to block list certain numbers (premium > ones). > I populated these numbers with in "globalblacklist" table. I followed > documentation: > > http://www.opensips.org/html/docs/modules/2.4.x/userblacklist.html#func_check_blacklist > > While part of destination number defined as prefix, like 1437, > blacklisting works and all the calls starting with destination number > '1437' will block. > id prefix whitelist description > 1 1437 0 NULL > > But for the required case, if I add full number in prefix column, like > 14375436291, opensips will not block the call. > id prefix whitelist description > 1 14375436291 0 NULL > > How can i block list of full numbers through opensips? > > -- > regards, > > abdul basit > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kkothari157 at gmail.com Mon Jul 16 01:01:57 2018 From: kkothari157 at gmail.com (Ketan Kothari) Date: Mon, 16 Jul 2018 10:31:57 +0530 Subject: [OpenSIPS-Users] SRTP & TLS on cluster (Opensips + Freeswitch) Message-ID: Hello All, I have configured Cluster setup using opensips + freeswitch. This is flow : UAC --> Opensips --> Freeswitch ---> OPensips --> UAS. Now i want to configure SRTP and TLS on cluster. What is best way to implement SRTP & TLS. ? Any suggestion ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From kurgan-rus at inbox.ru Mon Jul 16 04:41:55 2018 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Mon, 16 Jul 2018 11:41:55 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?SRTP_=26_TLS_on_cluster_=28Opensips_+_?= =?utf-8?q?Freeswitch=29?= In-Reply-To: References: Message-ID: <1531730515.712840711@f231.i.mail.ru> Hello I think this is a good starting point: http://www.opensips.org/Documentation/Tutorials#toc20 ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ From kkothari157 at gmail.com Mon Jul 16 04:53:05 2018 From: kkothari157 at gmail.com (Ketan Kothari) Date: Mon, 16 Jul 2018 14:23:05 +0530 Subject: [OpenSIPS-Users] SRTP & TLS on cluster (Opensips + Freeswitch) In-Reply-To: <1531730515.712840711@f231.i.mail.ru> References: <1531730515.712840711@f231.i.mail.ru> Message-ID: Hello Alexey, Thanks for your reply. Yes but i want suggestion which one is better for cluster ? 1. Configured SRTP and TLS on opensips server 2. Configured SRTP and TLS on freeswitch server. Can i choose second point and use opensips as only load balance ? Is that possible ? On Mon, Jul 16, 2018 at 2:11 PM, Alexey Kazantsev via Users < users at lists.opensips.org> wrote: > Hello > > I think this is a good starting point: > > http://www.opensips.org/Documentation/Tutorials#toc20 > > ----------------------------------------------- > BR, Alexey > http://alexeyka.zantsev.com/ > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kurgan-rus at inbox.ru Mon Jul 16 05:09:35 2018 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Mon, 16 Jul 2018 12:09:35 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?SRTP_=26_TLS_on_cluster_=28Opensips_+_?= =?utf-8?q?Freeswitch=29?= In-Reply-To: References: <1531730515.712840711@f231.i.mail.ru> Message-ID: <1531732175.25632309@f342.i.mail.ru> It depends on what do you want to secure. But I'm not sure that you want to have secure channel only between OpenSIPS and Freeswitch :) I think you need a secure channel between end users and OpenSIPS, as an edge point of your VoIP network, am I right?! ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ From razvan at opensips.org Mon Jul 16 07:55:19 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 16 Jul 2018 14:55:19 +0300 Subject: [OpenSIPS-Users] Proto WSS No Open TCP Connection after reinvite In-Reply-To: References: Message-ID: Hi, Sebastian! The re-invite probably generates a remote contact update. And if you don't "fix" the contact on re-invites and their 200 OK, you might end up with broken contacts in the dialog, thus sequential signaling will not work. I suggest you do two things to debug this: 1. remove the fix_route_dialog() call - the call should still be routed according to RR information, presuming this information is correct. 2. start the call, run `opensipsctl fifo dlg_list` and write down the WSS's contact, then put the call on hold, and check again the contact. Best regards, Răzvan On 07/13/2018 09:19 PM, Sebastian Sastre wrote: > > Hello, I’ve been experiencing a situation with Proto WSS. The scenario > is very simple. A call is established from an Asterisk Box to Opensips > (UDP) and finally a SipJs7.8 (WSS). Everything works great and we are > able to register using mid registrar and pass calls thru. > > When an agent puts the call on hold a reinvite is correctly negotiated > and the call is placed on hold and viceversa.   However!, if the > originating caller disconnects the call while still on hold, Asterisk > will correctly terminate the dialog with a Bye but when OpenSIPs will > complain about not finding a suitable tcp connection and responds with a > 477 even after successfully matching and processing the dialog > termination correctly. > > opensipsctl fifo list_tcp_conns  shows the connection available. > > The only way I found of fixing this problem is by adding > fix_route_dialog() on the sequential loose route. > > if (loose_route()) { > if (is_method("BYE")) { >                         if (!validate_dialog()){ >                               fix_route_dialog(); >                         } > > What do you guys think? > Am I messing up something in the script or is this the correct way to > address this problem? > > The funny thing is that there is no difference notable between the bye > after hold and a regular bye without putting the call on hold. > Here is the opensips log with the error and the trace. > > https://pastebin.com/BEJ6fAR8 > > Thanks ! > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From marketing at opensips.org Mon Jul 16 10:37:51 2018 From: marketing at opensips.org (OpenSIPS Team) Date: Mon, 16 Jul 2018 17:37:51 +0300 Subject: [OpenSIPS-Users] A New OpenSIPS Bootcamp training session Message-ID: <7b90aae7-2aee-8b0e-3d06-698bdde55dfe@opensips.org> * 22nd-26th of October 2018, Cluj-Napoca, Romania* *OpenSIPS Bootcamp!* Join the core developers and founder of the OpenSIPS project for a new *OpenSIPS Bootcamp* session in Europe - a five days (40 hours) intensive and practical training, covering installation, configuration and administration of *OpenSIPS.* This year's curriculum has been aligned to OpenSIPS 2.4, but also enhanced to cover some of the latest introduced features like OpenSIPS clustering. Join us in Cluj-Napoca, the Silicon Valley of Romania and one of the most beautiful and touristy places in Transylvania! Check the Schedule now *Early Birds open * The Early Bird 10% discount is available for registrations before /*1st of September*/, so do not mis the opportunity. The number of seats is limited, so be sure and book a seat now. Keep in mind that a 10% group discount is also available - grab your work mate and start learning more OpenSIPS together . Register Now! * * *Certified training saves time and money* OpenSIPS mistakes are easily avoided if you get proper training! Companies that use OpenSIPS waste time and money when they don't have a trained engineer on staff. Searching on Google, waiting on IRC, even the latency in mailing list replies takes it's toll over time. Take this rare opportunity to train your employees with the project members themselves. Get training! Any questions? do not hesitate to contact us ! ------------------------------------------------------------------------ You received this email as part of your relationship with the OpenSIPS Project. If you do not want to receive any more news, please email to unsubscribe . -------------- next part -------------- An HTML attachment was scrubbed... URL: From aymane.sabbane at gmail.com Mon Jul 16 12:39:18 2018 From: aymane.sabbane at gmail.com (Aymane Sabbane) Date: Mon, 16 Jul 2018 17:39:18 +0100 Subject: [OpenSIPS-Users] Unconditional call forward Message-ID: Hi Guys, I am trying to configure Unconditional call forward on my OpenSIPs machine using below lines on script: if(avp_db_load("$ru","$avp(callfwd)")){ $ru=$avp(s:callfwd); xlog("forwarded to $avp(callfwd)"); route(relay); exit; } I used after this "opensipctl avp add" and i was able to see my callfwd rule under usr_preferences database. When I call the extension that needs to be forwarded opensips challenges my invite twice, you can see it in the attached screenshot Do you guys have any idea why the server behaves like that? Best regards, *------------------------------------------------------* *SABBANE Aymane* VoIP & Network/Telecom Engineer Mobile: + (212) 06 62 14 45 66 *------------------------------------------------------* -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 4314 bytes Desc: not available URL: From tito at xsvoce.com Mon Jul 16 17:28:24 2018 From: tito at xsvoce.com (Tito Cumpen) Date: Mon, 16 Jul 2018 14:28:24 -0700 Subject: [OpenSIPS-Users] Docker centos In-Reply-To: References: Message-ID: For anyone who has to deal with this sort of scenario is added something to the init script within the start function. PUBLICIP=$(curl -s http://whatismyip.akamai.com/) echo $PUBLICIP sed -i '/^advertised_address/d' /etc/opensips/opensips.m4 sed -i '/####### Modules Section ########$/i\advertised_address=''\"'$PUBLICIP'\"' /etc/opensips/opensips.m4 # there is something at end of this output which is needed to # report proper [ OK ] status in Fedora scripts m4 /etc/opensips/defines.m4 /etc/opensips/opensips.m4 > /etc/opensips/opensips.cfg This takes the public ip returned by the curl command and modifies the .m4 or cfg script with it. On Thu, Jul 12, 2018 at 10:27 AM, Tito Cumpen wrote: > Hello, > > I've made an image of opensips in centos 7 but one of my concerns is that > docker uses internal IP addresses. With this being the case is there any > way to make opensips use stun to figure out its public IP? I tried > referencing the GitHub docker-opensips which runs in Debian but I cannot > find anything in there addresses this issue. > > Thanks, > Tito > -------------- next part -------------- An HTML attachment was scrubbed... URL: From k.galinurov at gmail.com Tue Jul 17 05:58:56 2018 From: k.galinurov at gmail.com (Kirill Galinurov) Date: Tue, 17 Jul 2018 12:58:56 +0300 Subject: [OpenSIPS-Users] mi_json returns invalid JSON Message-ID: Hi. I use opensips mi_json module to get statistics. But for some commands it returns incorrect JSON. For example /json/dr_reload_status returns {"Partition": [{"value":"sipprovider_def", "attributes":{"Date": "Mon Jul 16 07:10:58 2018 "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 07:11:59 2018 "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 07:11:59 2018 "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 07:12:54 2018 "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:12:54 2018 "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:13:51 2018 "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 07:13:52 2018 "}}]} This is incorrect, because it shoulde not include "\n" symbols. -------------- next part -------------- An HTML attachment was scrubbed... URL: From gerwin.van.de.steeg at vadacom.com Tue Jul 17 07:35:38 2018 From: gerwin.van.de.steeg at vadacom.com (Gerwin van de Steeg) Date: Tue, 17 Jul 2018 23:35:38 +1200 Subject: [OpenSIPS-Users] Multiple branches for a single AOR Message-ID: Folks, I'm trying to narrow down a 482 Merged Request problem on calls from one SIP device to another via OpenSIPS 2.4.1. Yealink T41P SIP device (A-party), calls via OpenSIPS, to another AOR owned by a Zoiper5 device (B-party). The intent is to ensure that when the B-party rejects the call with a 486 Busy Here, that the response code gets sent through to A-party. However what I'm seeing is the 486 gets sent to OpenSIPS which ACK's it, but doesn't go anywhere from there, and then something causes a second invite to be sent from OpenSIPS to the B-party which then responds of course with 482 Merged Request. The call as it is progressing through the call flow seems to be starting a second branch to the AOR (only one SIP device registered using UDP per AOR). What would be causing that second call so that I can eliminate it and get to the behaviour I'm expecting. Just using a slightly modified residential default config template with websocket support (the problem was noticed using SIP.JS but exists also in generic SIP device to SIP device calls). Image containing sngrep of call: https://imgur.com/RCZXkO6 Subscribers are in the form of @ With an alias setup for an extension number. ie. alfred.anderson at ... = 552 alice.bell at ... = 553 excerpt from opensips.cfg if ($rU==NULL) { # request with no Username in RURI send_reply("484","Address Incomplete"); exit; } $acc_extra(src_ip) = $si; # source IP of the request $acc_leg(caller) = $fu; $acc_leg(callee) = $ru; # apply DB based aliases if (alias_db_lookup("dbaliases")) { xlog("Alias lookup success [$fu/$tu/$ru/$ci]"); } else { xlog("Alias lookup failure [$fu/$tu/$ru/$ci]"); } # do blind callforward lookup if (avp_db_load("rU", "$avp(callfwd)")) { t_reply("181", "Call Is Being Forwarded"); $ru = $avp(callfwd); xlog("forwarded call to: $avp(callfwd)"); route(relay); exit; } # apply transformations from dialplan table dp_translate("0", "$rU/$rU"); # check if the call needs to be routed to freeswitch route(freeswitch); # here we would set the redirect URI if it had one route(lookup); } route[lookup] { script_trace(1, "$rm from $si, rur=$ru", "me"); xlog("route:lookup"); # do lookup with method filtering if (!lookup("location","m")) { xlog("lookup failure"); t_newtran(); if (!db_does_uri_exist()) { xlog("$cfg_line: URI doesn't exist"); send_reply("420", "Bad Extension"); exit; } t_reply("404", "Not Found"); exit; } # when routing via usrloc, log the missed calls also do_accounting("db","missed"); route(relay); } route[freeswitch] { xlog("route:freeswitch"); if (!is_method("INVITE")) { return; } # if the called number begins with the right dialplan redirect it to freeswitch # here we take everythign prefixed with a *, strip it, and send it to freeswitch if ($rU=~"^\*") { strip(1); $du = "sip:10.23.4.192:50600"; route(relay); } } route[relay] { xlog("route:relay: Relaying: method=$rm"); # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("per_branch_ops"); t_on_reply("handle_nat"); t_on_failure("missed_call"); } else if (is_method("BYE|CANCEL")) { # cancel the rtpengine transcoding rtpengine_delete(); } if (!t_relay()) { send_reply("500","Internal Error"); } exit; } branch_route[per_branch_ops] { script_trace(1, "$rm from $si, rur=$ru", "me"); xlog("[$ci/$T_branch_idx] branch_route:per_branch_ops: new branch at $ru\n"); # WebSocket specific handling with NORMAL SDP negotiation # assumes SDP offer in the INVITE from the UAC, and SDP # answer is in 200 OK from the UAS if (!is_method("INVITE") || !has_body("application/sdp")) return; if (isflagset(SRC_WS) && isbflagset(DST_WS)) $var(rtpengine_flags) = "ICE=force-relay DTLS=passive"; else if (isflagset(SRC_WS) && !isbflagset(DST_WS)) $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; else if (!isflagset(SRC_WS) && isbflagset(DST_WS)) $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; else if (!isflagset(SRC_WS) && !isbflagset(DST_WS)) $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; # only enable transcoding if websocket call for now if (isflagset(SRC_WS) || isbflagset(DST_WS)) { rtpengine_offer("$var(rtpengine_flags)"); } } onreply_route[handle_nat] { script_trace(1, "$rm from $si, rur=$ru", "me"); xlog("[$ci/$T_branch_idx] onreply_route:handle_nat: $ru\n"); # WebSocket specific handling with NORMAL SDP negotiation # assumes SDP offer in the INVITE from the UAC, and SDP # answer is in 200 OK from the UAS if (!has_body("application/sdp")) return; if (isflagset(SRC_WS) && isbflagset(DST_WS)) $var(rtpengine_flags) = "ICE=force-relay DTLS=passive"; else if (isflagset(SRC_WS) && !isbflagset(DST_WS)) $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; else if (!isflagset(SRC_WS) && isbflagset(DST_WS)) $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; else if (!isflagset(SRC_WS) && !isbflagset(DST_WS)) $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; # only enable transcoding if websocket call for now if (isflagset(SRC_WS) || isbflagset(DST_WS)) { rtpengine_answer("$var(rtpengine_flags)"); } } failure_route[missed_call] { script_trace(1, "$rm from $si, rur=$ru", "me"); xlog("[$ci/$T_branch_idx] failure_route:missed_call: incoming failure response to $rm <- $T_reply_code/$T_ruri"); if (t_was_cancelled()) { xlog("[$ci/$T_branch_idx] was cancelled"); exit; } do_accounting("db", "missed"); if (!t_relay()) { send_reply("500","Internal Error"); } else { xlog("[$ci/$T_branch_idx] Relay success $rm/$T_reply_code"); } } Cheers, Gerwin -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Tue Jul 17 08:08:53 2018 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Tue, 17 Jul 2018 09:08:53 -0300 Subject: [OpenSIPS-Users] soket error In-Reply-To: <218785706.1664377.1531232227011@mail.yahoo.com> References: <1531158547.3207.7@smtp.networklab.ca> <1882020822.1561348.1531215889633@mail.yahoo.com> <1531221654.19705.0@smtp.networklab.ca> <218785706.1664377.1531232227011@mail.yahoo.com> Message-ID: <1531829333.2178.5@smtp.networklab.ca> Hello Everyone, Anybody can give insight ? volga629. On Tue, Jul 10, 2018 at 11:17 AM, Pasan Meemaduma wrote: > HI Volga, > > I haven't used mhomed=1 param, Apologies not sure how to help there. > > > On Tuesday, 10 July 2018, 4:51:12 PM GMT+5:30, > wrote: > > > Hello Pasan, > The whole issue is mhomed=1 it can't determine correct socket or don't > have logic to work with vip addresses. I have to use on LAN side > > force_send_socket(udp:vip1:5060); > > But issue right now if one node goes down and vip is relocated to > another node, how to determine which socket to use to send call from > correct soure ip ? > > > volga629 > > On Tue, Jul 10, 2018 at 6:44 AM, Pasan Meemaduma via Users > wrote: > > Hi Volga, > > > > Its a very common question about non local address binding in linux. > > did you set following via sysctl ? > > net.ipv4.ip_nonlocal_bind=1 > > > > > > > > On Monday, 9 July 2018, 11:20:04 PM GMT+5:30, volga629 at networklab.ca > > wrote: > > > > > > Hello Everyone, > > Trying build test cluster active/active with 3 vips for each node > > separate virtual ip with keepalived > > And opensips failing to bound vip ip for corresponding vm > > This is interface config with vip > > > > 4: ens37: mtu 1500 qdisc fq state > > UNKNOWN group default qlen 1000 > > link/ether 00:50:56:a2:d5:0c brd ff:ff:ff:ff:ff:ff > > inet 10.100.104.4/28 brd 10.100.104.15 scope global ens37 > > valid_lft forever preferred_lft forever > > inet 10.100.104.7/28 scope global secondary ens37 > > ---> Keepalived VIP > > valid_lft forever preferred_lft forever > > inet6 fe80::b279:d4d6:45a1:6dd6/64 scope link > > valid_lft forever preferred_lft forever > > inet6 fe80::f9a7:9cc4:6a27:4cde/64 scope link dadfailed tentative > > valid_lft forever preferred_lft forever > > > > > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: Forwarding REGISTER > to > > main registrar ~> [] > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:core:get_out_socket: no socket found > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:mid_registrar:uri2sock: no corresponding socket for af 2 > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:mid_registrar:overwrite_req_contacts: failed to obtain next > hop > > socket, ci=0_3830065743 at 192.168.1.13 > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:mid_registrar:mid_reg_req_fwded: failed to overwrite Contact > > URIs > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:core:get_out_socket: no socket found > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 1 (no > > corresponding listening socket) > > Jul 9 12:34:42 sbc01 /usr/sbin/opensips[6019]: > > ERROR:tm:t_forward_nonack: failure to add branches > > > > > > Listen > > > > listen=udp:10.100.104.7:5060 vip1 > > listen=udp:10.100.104.8:5060 vip2 > > listen=udp:10.100.104.9:5060 vip3 > > listen=bin:10.100.104.7:5585 bin cluster > > > > Any idea what what will be possible cause ? > > > > volga629 > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From liviu at opensips.org Tue Jul 17 08:48:58 2018 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 17 Jul 2018 15:48:58 +0300 Subject: [OpenSIPS-Users] Multiple branches for a single AOR In-Reply-To: References: Message-ID: Hi Gerwin, Inside your failure route, you are always attempting a retry of any failed request.  This logic conflicts with your initial statement that "The intent is to ensure that the response code gets sent through to A-party". Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 17.07.2018 14:35, Gerwin van de Steeg wrote: > Folks, > > I'm trying to narrow down a 482 Merged Request problem on calls from > one SIP device to another via OpenSIPS 2.4.1. Yealink T41P SIP device > (A-party), calls via OpenSIPS, to another AOR owned by a Zoiper5 > device (B-party). > The intent is to ensure that when the B-party rejects the call with a > 486 Busy Here, that the response code gets sent through to A-party.  > However what I'm seeing is the 486 gets sent to OpenSIPS which ACK's > it, but doesn't go anywhere from there, and then something causes a > second invite to be sent from OpenSIPS to the B-party which then > responds of course with 482 Merged Request. > The call as it is progressing through the call flow seems to be > starting a second branch to the AOR (only one SIP device registered > using UDP per AOR). > > What would be causing that second call so that I can eliminate it and > get to the behaviour I'm expecting.  Just using a slightly modified > residential default config template with websocket support (the > problem was noticed using SIP.JS but exists also in generic SIP device > to SIP device calls). > > Image containing sngrep of call: https://imgur.com/RCZXkO6 > > Subscribers are in the form of @ > With an alias setup for an extension number. > > ie. >   alfred.anderson at ... = 552 > alice.bell at ... = 553 > > excerpt from opensips.cfg > >        if ($rU==NULL) { >                # request with no Username in RURI >                send_reply("484","Address Incomplete"); >                exit; >        } > >        $acc_extra(src_ip) = $si; # source IP of the request >        $acc_leg(caller) = $fu; >        $acc_leg(callee) = $ru; > >        # apply DB based aliases >        if (alias_db_lookup("dbaliases")) { >                xlog("Alias lookup success [$fu/$tu/$ru/$ci]"); >        } >        else { >                xlog("Alias lookup failure [$fu/$tu/$ru/$ci]"); >        } > >        # do blind callforward lookup >        if (avp_db_load("rU", "$avp(callfwd)")) { >                t_reply("181", "Call Is Being Forwarded"); >                $ru = $avp(callfwd); >                xlog("forwarded call to: $avp(callfwd)"); >                route(relay); >                exit; >        } > >        # apply transformations from dialplan table >        dp_translate("0", "$rU/$rU"); > >        # check if the call needs to be routed to freeswitch >        route(freeswitch); > >        # here we would set the redirect URI if it had one >        route(lookup); > } > > route[lookup] { >      script_trace(1, "$rm from $si, rur=$ru", "me"); >        xlog("route:lookup"); >        # do lookup with method filtering >        if (!lookup("location","m")) { >                xlog("lookup failure"); >                t_newtran(); >                if (!db_does_uri_exist()) { >                        xlog("$cfg_line: URI doesn't exist"); >                        send_reply("420", "Bad Extension"); >                        exit; >                } >                t_reply("404", "Not Found"); >                exit; >        } > >        # when routing via usrloc, log the missed calls also >        do_accounting("db","missed"); > >        route(relay); > } > > route[freeswitch] { >        xlog("route:freeswitch"); >        if (!is_method("INVITE")) { >                return; >        } > >        # if the called number begins with the right dialplan redirect > it to freeswitch >        # here we take everythign prefixed with a *, strip it, and send > it to freeswitch >        if ($rU=~"^\*") { >                strip(1); >                $du = "sip:10.23.4.192:50600 "; >                route(relay); >        } > } > > > route[relay] { >        xlog("route:relay: Relaying: method=$rm"); >        # for INVITEs enable some additional helper routes >        if (is_method("INVITE")) { >                t_on_branch("per_branch_ops"); >                t_on_reply("handle_nat"); >                t_on_failure("missed_call"); >        } >        else if (is_method("BYE|CANCEL")) { >                # cancel the rtpengine transcoding >                rtpengine_delete(); >        } > >        if (!t_relay()) { >                send_reply("500","Internal Error"); >        } >        exit; > } > > > branch_route[per_branch_ops] { >        script_trace(1, "$rm from $si, rur=$ru", "me"); >        xlog("[$ci/$T_branch_idx] branch_route:per_branch_ops: new > branch at $ru\n"); > >        # WebSocket specific handling with NORMAL SDP negotiation >        # assumes SDP offer in the INVITE from the UAC, and SDP >        # answer is in 200 OK from the UAS >        if (!is_method("INVITE") || !has_body("application/sdp")) >                return; > >        if (isflagset(SRC_WS) && isbflagset(DST_WS)) >                $var(rtpengine_flags) = "ICE=force-relay DTLS=passive"; >        else if (isflagset(SRC_WS) && !isbflagset(DST_WS)) >                $var(rtpengine_flags) = "RTP/AVP > replace-session-connection replace-origin ICE=remove"; >        else if (!isflagset(SRC_WS) && isbflagset(DST_WS)) >                $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >        else if (!isflagset(SRC_WS) && !isbflagset(DST_WS)) >                $var(rtpengine_flags) = "RTP/AVP > replace-session-connection replace-origin ICE=remove"; > >        # only enable transcoding if websocket call for now >        if (isflagset(SRC_WS) || isbflagset(DST_WS)) { >            rtpengine_offer("$var(rtpengine_flags)"); >        } > } > > onreply_route[handle_nat] { >        script_trace(1, "$rm from $si, rur=$ru", "me"); >        xlog("[$ci/$T_branch_idx] onreply_route:handle_nat: $ru\n"); > >        # WebSocket specific handling with NORMAL SDP negotiation >        # assumes SDP offer in the INVITE from the UAC, and SDP >        # answer is in 200 OK from the UAS >        if (!has_body("application/sdp")) >                return; > >        if (isflagset(SRC_WS) && isbflagset(DST_WS)) >                $var(rtpengine_flags) = "ICE=force-relay DTLS=passive"; >        else if (isflagset(SRC_WS) && !isbflagset(DST_WS)) >                $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >        else if (!isflagset(SRC_WS) && isbflagset(DST_WS)) >                $var(rtpengine_flags) = "RTP/AVP > replace-session-connection replace-origin ICE=remove"; >        else if (!isflagset(SRC_WS) && !isbflagset(DST_WS)) >                $var(rtpengine_flags) = "RTP/AVP > replace-session-connection replace-origin ICE=remove"; > >        # only enable transcoding if websocket call for now >        if (isflagset(SRC_WS) || isbflagset(DST_WS)) { >                rtpengine_answer("$var(rtpengine_flags)"); >        } > } > > failure_route[missed_call] { >        script_trace(1, "$rm from $si, rur=$ru", "me"); >        xlog("[$ci/$T_branch_idx] failure_route:missed_call: incoming > failure response to $rm <- $T_reply_code/$T_ruri"); >        if (t_was_cancelled()) { >                xlog("[$ci/$T_branch_idx] was cancelled"); >                exit; >        } >        do_accounting("db", "missed"); > >        if (!t_relay()) { >                send_reply("500","Internal Error"); >        } >        else { >                xlog("[$ci/$T_branch_idx] Relay success > $rm/$T_reply_code"); >        } > } > > > > > > Cheers, >    Gerwin > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Jul 17 08:54:14 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 17 Jul 2018 12:54:14 +0000 Subject: [OpenSIPS-Users] Multiple branches for a single AOR In-Reply-To: References: Message-ID: <264A3805-19AF-4DF3-BB74-BFA0E490FC19@genesys.com> Gerwin, Specifically, it is the t_relay() call within your failure_route. t_relay() is used to send requests, not responses. The automatic action of failure_route if no retransmission is attempted is to send the reply back upstream [1]. So you don’t have to do anything if that is the functionality you desire. [1] http://www.opensips.org/Documentation/Script-Routes-2-4#toc3 Ben Newlin From: Users on behalf of Liviu Chircu Reply-To: OpenSIPS users mailling list Date: Tuesday, July 17, 2018 at 8:50 AM To: "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] Multiple branches for a single AOR Hi Gerwin, Inside your failure route, you are always attempting a retry of any failed request. This logic conflicts with your initial statement that "The intent is to ensure that the response code gets sent through to A-party". Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 17.07.2018 14:35, Gerwin van de Steeg wrote: Folks, I'm trying to narrow down a 482 Merged Request problem on calls from one SIP device to another via OpenSIPS 2.4.1. Yealink T41P SIP device (A-party), calls via OpenSIPS, to another AOR owned by a Zoiper5 device (B-party). The intent is to ensure that when the B-party rejects the call with a 486 Busy Here, that the response code gets sent through to A-party. However what I'm seeing is the 486 gets sent to OpenSIPS which ACK's it, but doesn't go anywhere from there, and then something causes a second invite to be sent from OpenSIPS to the B-party which then responds of course with 482 Merged Request. The call as it is progressing through the call flow seems to be starting a second branch to the AOR (only one SIP device registered using UDP per AOR). What would be causing that second call so that I can eliminate it and get to the behaviour I'm expecting. Just using a slightly modified residential default config template with websocket support (the problem was noticed using SIP.JS but exists also in generic SIP device to SIP device calls). Image containing sngrep of call: https://imgur.com/RCZXkO6 Subscribers are in the form of @ With an alias setup for an extension number. ie. alfred.anderson at ... = 552 alice.bell at ... = 553 excerpt from opensips.cfg if ($rU==NULL) { # request with no Username in RURI send_reply("484","Address Incomplete"); exit; } $acc_extra(src_ip) = $si; # source IP of the request $acc_leg(caller) = $fu; $acc_leg(callee) = $ru; # apply DB based aliases if (alias_db_lookup("dbaliases")) { xlog("Alias lookup success [$fu/$tu/$ru/$ci]"); } else { xlog("Alias lookup failure [$fu/$tu/$ru/$ci]"); } # do blind callforward lookup if (avp_db_load("rU", "$avp(callfwd)")) { t_reply("181", "Call Is Being Forwarded"); $ru = $avp(callfwd); xlog("forwarded call to: $avp(callfwd)"); route(relay); exit; } # apply transformations from dialplan table dp_translate("0", "$rU/$rU"); # check if the call needs to be routed to freeswitch route(freeswitch); # here we would set the redirect URI if it had one route(lookup); } route[lookup] { script_trace(1, "$rm from $si, rur=$ru", "me"); xlog("route:lookup"); # do lookup with method filtering if (!lookup("location","m")) { xlog("lookup failure"); t_newtran(); if (!db_does_uri_exist()) { xlog("$cfg_line: URI doesn't exist"); send_reply("420", "Bad Extension"); exit; } t_reply("404", "Not Found"); exit; } # when routing via usrloc, log the missed calls also do_accounting("db","missed"); route(relay); } route[freeswitch] { xlog("route:freeswitch"); if (!is_method("INVITE")) { return; } # if the called number begins with the right dialplan redirect it to freeswitch # here we take everythign prefixed with a *, strip it, and send it to freeswitch if ($rU=~"^\*") { strip(1); $du = "sip:10.23.4.192:50600"; route(relay); } } route[relay] { xlog("route:relay: Relaying: method=$rm"); # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("per_branch_ops"); t_on_reply("handle_nat"); t_on_failure("missed_call"); } else if (is_method("BYE|CANCEL")) { # cancel the rtpengine transcoding rtpengine_delete(); } if (!t_relay()) { send_reply("500","Internal Error"); } exit; } branch_route[per_branch_ops] { script_trace(1, "$rm from $si, rur=$ru", "me"); xlog("[$ci/$T_branch_idx] branch_route:per_branch_ops: new branch at $ru\n"); # WebSocket specific handling with NORMAL SDP negotiation # assumes SDP offer in the INVITE from the UAC, and SDP # answer is in 200 OK from the UAS if (!is_method("INVITE") || !has_body("application/sdp")) return; if (isflagset(SRC_WS) && isbflagset(DST_WS)) $var(rtpengine_flags) = "ICE=force-relay DTLS=passive"; else if (isflagset(SRC_WS) && !isbflagset(DST_WS)) $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; else if (!isflagset(SRC_WS) && isbflagset(DST_WS)) $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; else if (!isflagset(SRC_WS) && !isbflagset(DST_WS)) $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; # only enable transcoding if websocket call for now if (isflagset(SRC_WS) || isbflagset(DST_WS)) { rtpengine_offer("$var(rtpengine_flags)"); } } onreply_route[handle_nat] { script_trace(1, "$rm from $si, rur=$ru", "me"); xlog("[$ci/$T_branch_idx] onreply_route:handle_nat: $ru\n"); # WebSocket specific handling with NORMAL SDP negotiation # assumes SDP offer in the INVITE from the UAC, and SDP # answer is in 200 OK from the UAS if (!has_body("application/sdp")) return; if (isflagset(SRC_WS) && isbflagset(DST_WS)) $var(rtpengine_flags) = "ICE=force-relay DTLS=passive"; else if (isflagset(SRC_WS) && !isbflagset(DST_WS)) $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; else if (!isflagset(SRC_WS) && isbflagset(DST_WS)) $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; else if (!isflagset(SRC_WS) && !isbflagset(DST_WS)) $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; # only enable transcoding if websocket call for now if (isflagset(SRC_WS) || isbflagset(DST_WS)) { rtpengine_answer("$var(rtpengine_flags)"); } } failure_route[missed_call] { script_trace(1, "$rm from $si, rur=$ru", "me"); xlog("[$ci/$T_branch_idx] failure_route:missed_call: incoming failure response to $rm <- $T_reply_code/$T_ruri"); if (t_was_cancelled()) { xlog("[$ci/$T_branch_idx] was cancelled"); exit; } do_accounting("db", "missed"); if (!t_relay()) { send_reply("500","Internal Error"); } else { xlog("[$ci/$T_branch_idx] Relay success $rm/$T_reply_code"); } } Cheers, Gerwin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Tue Jul 17 09:02:50 2018 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 17 Jul 2018 16:02:50 +0300 Subject: [OpenSIPS-Users] Multiple branches for a single AOR In-Reply-To: <264A3805-19AF-4DF3-BB74-BFA0E490FC19@genesys.com> References: <264A3805-19AF-4DF3-BB74-BFA0E490FC19@genesys.com> Message-ID: <64fcb3d3-611e-db28-328b-8aee91def2c9@opensips.org> Hi Ben, Just for the sake of completeness, if we do decide to retry a request within a failure route, we will not do a "retransmission", but actually create a brand new transaction (with a unique ";branch=" param)on the SIP proxy's UAC side (the outgoing side). Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 17.07.2018 15:54, Ben Newlin wrote: > > Gerwin, > > Specifically, it is the t_relay() call within your failure_route. > t_relay()  is used to send requests, not responses. The automatic > action of failure_route if no retransmission is attempted is to send > the reply back upstream [1]. So you don’t have to do anything if that > is the functionality you desire. > > [1] http://www.opensips.org/Documentation/Script-Routes-2-4#toc3 > > > Ben Newlin > > *From: *Users on behalf of Liviu > Chircu > *Reply-To: *OpenSIPS users mailling list > *Date: *Tuesday, July 17, 2018 at 8:50 AM > *To: *"users at lists.opensips.org" > *Subject: *Re: [OpenSIPS-Users] Multiple branches for a single AOR > > Hi Gerwin, > > Inside your failure route, you are always attempting a retry of any > failed request. This logic conflicts with your initial statement that > "The intent is to ensure that the response code gets sent through to > A-party". > > Best regards, > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 17.07.2018 14:35, Gerwin van de Steeg wrote: > > Folks, > > I'm trying to narrow down a 482 Merged Request problem on calls > from one SIP device to another via OpenSIPS 2.4.1.  Yealink T41P > SIP device (A-party), calls via OpenSIPS, to another AOR owned by > a Zoiper5 device (B-party). > > The intent is to ensure that when the B-party rejects the call > with a 486 Busy Here, that the response code gets sent through to > A-party.  However what I'm seeing is the 486 gets sent to OpenSIPS > which ACK's it, but doesn't go anywhere from there, and then > something causes a second invite to be sent from OpenSIPS to the > B-party which then responds of course with 482 Merged Request. > > The call as it is progressing through the call flow seems to be > starting a second branch to the AOR (only one SIP device > registered using UDP per AOR). > > What would be causing that second call so that I can eliminate it > and get to the behaviour I'm expecting.  Just using a slightly > modified residential default config template with websocket > support (the problem was noticed using SIP.JS but exists also in > generic SIP device to SIP device calls). > > Image containing sngrep of call: https://imgur.com/RCZXkO6 > > > Subscribers are in the form of @ > With an alias setup for an extension number. > > ie. >   alfred.anderson at ... = 552 > > alice.bell at ... = 553 > > excerpt from opensips.cfg > >        if ($rU==NULL) { >                # request with no Username in RURI >                send_reply("484","Address Incomplete"); >                exit; >        } > >        $acc_extra(src_ip) = $si; # source IP of the request >        $acc_leg(caller) = $fu; >        $acc_leg(callee) = $ru; > >        # apply DB based aliases >        if (alias_db_lookup("dbaliases")) { >                xlog("Alias lookup success [$fu/$tu/$ru/$ci]"); >        } >        else { >                xlog("Alias lookup failure [$fu/$tu/$ru/$ci]"); >        } > >        # do blind callforward lookup >        if (avp_db_load("rU", "$avp(callfwd)")) { >                t_reply("181", "Call Is Being Forwarded"); >                $ru = $avp(callfwd); >                xlog("forwarded call to: $avp(callfwd)"); >                route(relay); >                exit; >        } > >        # apply transformations from dialplan table >        dp_translate("0", "$rU/$rU"); > >        # check if the call needs to be routed to freeswitch >        route(freeswitch); > >        # here we would set the redirect URI if it had one >        route(lookup); > } > > route[lookup] { >        script_trace(1, "$rm from $si, rur=$ru", "me"); >      xlog("route:lookup"); >        # do lookup with method filtering >        if (!lookup("location","m")) { >                xlog("lookup failure"); >                t_newtran(); >                if (!db_does_uri_exist()) { >                        xlog("$cfg_line: URI doesn't exist"); >                        send_reply("420", "Bad Extension"); >                        exit; >                } >                t_reply("404", "Not Found"); >                exit; >        } > >        # when routing via usrloc, log the missed calls also >        do_accounting("db","missed"); > >        route(relay); > } > > route[freeswitch] { >        xlog("route:freeswitch"); >        if (!is_method("INVITE")) { >                return; >        } > >        # if the called number begins with the right dialplan > redirect it to freeswitch >        # here we take everythign prefixed with a *, strip it, and > send it to freeswitch >        if ($rU=~"^\*") { >                strip(1); >                $du = "sip:10.23.4.192:50600 > "; >                route(relay); >        } > } > > > route[relay] { >        xlog("route:relay: Relaying: method=$rm"); >        # for INVITEs enable some additional helper routes >        if (is_method("INVITE")) { >                t_on_branch("per_branch_ops"); >                t_on_reply("handle_nat"); >                t_on_failure("missed_call"); >        } >        else if (is_method("BYE|CANCEL")) { >                # cancel the rtpengine transcoding >                rtpengine_delete(); >        } > >        if (!t_relay()) { >                send_reply("500","Internal Error"); >        } >        exit; > } > > > branch_route[per_branch_ops] { >        script_trace(1, "$rm from $si, rur=$ru", "me"); >        xlog("[$ci/$T_branch_idx] branch_route:per_branch_ops: new > branch at $ru\n"); > >        # WebSocket specific handling with NORMAL SDP negotiation >        # assumes SDP offer in the INVITE from the UAC, and SDP >        # answer is in 200 OK from the UAS >        if (!is_method("INVITE") || !has_body("application/sdp")) >                return; > >        if (isflagset(SRC_WS) && isbflagset(DST_WS)) >                $var(rtpengine_flags) = "ICE=force-relay DTLS=passive"; >        else if (isflagset(SRC_WS) && !isbflagset(DST_WS)) >                $var(rtpengine_flags) = "RTP/AVP > replace-session-connection replace-origin ICE=remove"; >        else if (!isflagset(SRC_WS) && isbflagset(DST_WS)) >                $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >        else if (!isflagset(SRC_WS) && !isbflagset(DST_WS)) >                $var(rtpengine_flags) = "RTP/AVP > replace-session-connection replace-origin ICE=remove"; > >        # only enable transcoding if websocket call for now >        if (isflagset(SRC_WS) || isbflagset(DST_WS)) { >            rtpengine_offer("$var(rtpengine_flags)"); >        } > } > > onreply_route[handle_nat] { >        script_trace(1, "$rm from $si, rur=$ru", "me"); >        xlog("[$ci/$T_branch_idx] onreply_route:handle_nat: $ru\n"); > >        # WebSocket specific handling with NORMAL SDP negotiation >        # assumes SDP offer in the INVITE from the UAC, and SDP >        # answer is in 200 OK from the UAS >        if (!has_body("application/sdp")) >                return; > >        if (isflagset(SRC_WS) && isbflagset(DST_WS)) >                $var(rtpengine_flags) = "ICE=force-relay DTLS=passive"; >        else if (isflagset(SRC_WS) && !isbflagset(DST_WS)) >                $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; >        else if (!isflagset(SRC_WS) && isbflagset(DST_WS)) >                $var(rtpengine_flags) = "RTP/AVP > replace-session-connection replace-origin ICE=remove"; >        else if (!isflagset(SRC_WS) && !isbflagset(DST_WS)) >                $var(rtpengine_flags) = "RTP/AVP > replace-session-connection replace-origin ICE=remove"; > >        # only enable transcoding if websocket call for now >        if (isflagset(SRC_WS) || isbflagset(DST_WS)) { >  rtpengine_answer("$var(rtpengine_flags)"); >        } > } > > failure_route[missed_call] { >        script_trace(1, "$rm from $si, rur=$ru", "me"); >        xlog("[$ci/$T_branch_idx] failure_route:missed_call: > incoming failure response to $rm <- $T_reply_code/$T_ruri"); >        if (t_was_cancelled()) { >                xlog("[$ci/$T_branch_idx] was cancelled"); >                exit; >        } >        do_accounting("db", "missed"); > >        if (!t_relay()) { >                send_reply("500","Internal Error"); >        } >        else { >                xlog("[$ci/$T_branch_idx] Relay success > $rm/$T_reply_code"); >        } > } > > > Cheers, >    Gerwin > > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From gerwin.van.de.steeg at vadacom.com Tue Jul 17 09:27:05 2018 From: gerwin.van.de.steeg at vadacom.com (Gerwin van de Steeg) Date: Wed, 18 Jul 2018 01:27:05 +1200 Subject: [OpenSIPS-Users] Multiple branches for a single AOR In-Reply-To: <64fcb3d3-611e-db28-328b-8aee91def2c9@opensips.org> References: <264A3805-19AF-4DF3-BB74-BFA0E490FC19@genesys.com> <64fcb3d3-611e-db28-328b-8aee91def2c9@opensips.org> Message-ID: Perfect, thanks that fixed it. On 18 July 2018 at 01:02, Liviu Chircu wrote: > Hi Ben, > > Just for the sake of completeness, if we do decide to retry a request > within a failure route, we will not do a "retransmission", but actually > create a brand new transaction (with a unique ";branch=" param) on the > SIP proxy's UAC side (the outgoing side). > > Best regards, > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 17.07.2018 15:54, Ben Newlin wrote: > > Gerwin, > > > > Specifically, it is the t_relay() call within your failure_route. > t_relay() is used to send requests, not responses. The automatic action of > failure_route if no retransmission is attempted is to send the reply back > upstream [1]. So you don’t have to do anything if that is the functionality > you desire. > > > > [1] http://www.opensips.org/Documentation/Script-Routes-2-4#toc3 > > > > Ben Newlin > > > > *From: *Users > on behalf of Liviu Chircu > > *Reply-To: *OpenSIPS users mailling list > > *Date: *Tuesday, July 17, 2018 at 8:50 AM > *To: *"users at lists.opensips.org" > > *Subject: *Re: [OpenSIPS-Users] Multiple branches for a single AOR > > > > Hi Gerwin, > > Inside your failure route, you are always attempting a retry of any failed > request. This logic conflicts with your initial statement that "The intent > is to ensure that the response code gets sent through to A-party". > > Best regards, > > Liviu Chircu > > OpenSIPS Developer > > http://www.opensips-solutions.com > > On 17.07.2018 14:35, Gerwin van de Steeg wrote: > > Folks, > > > > I'm trying to narrow down a 482 Merged Request problem on calls from one > SIP device to another via OpenSIPS 2.4.1. Yealink T41P SIP device > (A-party), calls via OpenSIPS, to another AOR owned by a Zoiper5 device > (B-party). > > The intent is to ensure that when the B-party rejects the call with a 486 > Busy Here, that the response code gets sent through to A-party. However > what I'm seeing is the 486 gets sent to OpenSIPS which ACK's it, but > doesn't go anywhere from there, and then something causes a second invite > to be sent from OpenSIPS to the B-party which then responds of course with > 482 Merged Request. > > The call as it is progressing through the call flow seems to be starting a > second branch to the AOR (only one SIP device registered using UDP per AOR). > > > > What would be causing that second call so that I can eliminate it and get > to the behaviour I'm expecting. Just using a slightly modified residential > default config template with websocket support (the problem was noticed > using SIP.JS but exists also in generic SIP device to SIP device calls). > > > > Image containing sngrep of call: https://imgur.com/RCZXkO6 > > > > Subscribers are in the form of @ > With an alias setup for an extension number. > > > > ie. > alfred.anderson at ... = 552 > > alice.bell at ... = 553 > > > > excerpt from opensips.cfg > > > > if ($rU==NULL) { > # request with no Username in RURI > send_reply("484","Address Incomplete"); > exit; > } > > $acc_extra(src_ip) = $si; # source IP of the request > $acc_leg(caller) = $fu; > $acc_leg(callee) = $ru; > > # apply DB based aliases > if (alias_db_lookup("dbaliases")) { > xlog("Alias lookup success [$fu/$tu/$ru/$ci]"); > } > else { > xlog("Alias lookup failure [$fu/$tu/$ru/$ci]"); > } > > # do blind callforward lookup > if (avp_db_load("rU", "$avp(callfwd)")) { > t_reply("181", "Call Is Being Forwarded"); > $ru = $avp(callfwd); > xlog("forwarded call to: $avp(callfwd)"); > route(relay); > exit; > } > > # apply transformations from dialplan table > dp_translate("0", "$rU/$rU"); > > # check if the call needs to be routed to freeswitch > route(freeswitch); > > # here we would set the redirect URI if it had one > route(lookup); > } > > route[lookup] { > script_trace(1, "$rm from $si, rur=$ru", "me"); > xlog("route:lookup"); > # do lookup with method filtering > if (!lookup("location","m")) { > xlog("lookup failure"); > t_newtran(); > if (!db_does_uri_exist()) { > xlog("$cfg_line: URI doesn't exist"); > send_reply("420", "Bad Extension"); > exit; > } > t_reply("404", "Not Found"); > exit; > } > > # when routing via usrloc, log the missed calls also > do_accounting("db","missed"); > > route(relay); > } > > route[freeswitch] { > xlog("route:freeswitch"); > if (!is_method("INVITE")) { > return; > } > > # if the called number begins with the right dialplan redirect it > to freeswitch > # here we take everythign prefixed with a *, strip it, and send it > to freeswitch > if ($rU=~"^\*") { > strip(1); > $du = "sip:10.23.4.192:50600"; > route(relay); > } > } > > > route[relay] { > xlog("route:relay: Relaying: method=$rm"); > # for INVITEs enable some additional helper routes > if (is_method("INVITE")) { > t_on_branch("per_branch_ops"); > t_on_reply("handle_nat"); > t_on_failure("missed_call"); > } > else if (is_method("BYE|CANCEL")) { > # cancel the rtpengine transcoding > rtpengine_delete(); > } > > if (!t_relay()) { > send_reply("500","Internal Error"); > } > exit; > } > > > branch_route[per_branch_ops] { > script_trace(1, "$rm from $si, rur=$ru", "me"); > xlog("[$ci/$T_branch_idx] branch_route:per_branch_ops: new branch > at $ru\n"); > > # WebSocket specific handling with NORMAL SDP negotiation > # assumes SDP offer in the INVITE from the UAC, and SDP > # answer is in 200 OK from the UAS > if (!is_method("INVITE") || !has_body("application/sdp")) > return; > > if (isflagset(SRC_WS) && isbflagset(DST_WS)) > $var(rtpengine_flags) = "ICE=force-relay DTLS=passive"; > else if (isflagset(SRC_WS) && !isbflagset(DST_WS)) > $var(rtpengine_flags) = "RTP/AVP replace-session-connection > replace-origin ICE=remove"; > else if (!isflagset(SRC_WS) && isbflagset(DST_WS)) > $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; > else if (!isflagset(SRC_WS) && !isbflagset(DST_WS)) > $var(rtpengine_flags) = "RTP/AVP replace-session-connection > replace-origin ICE=remove"; > > # only enable transcoding if websocket call for now > if (isflagset(SRC_WS) || isbflagset(DST_WS)) { > rtpengine_offer("$var(rtpengine_flags)"); > } > } > > onreply_route[handle_nat] { > script_trace(1, "$rm from $si, rur=$ru", "me"); > xlog("[$ci/$T_branch_idx] onreply_route:handle_nat: $ru\n"); > > # WebSocket specific handling with NORMAL SDP negotiation > # assumes SDP offer in the INVITE from the UAC, and SDP > # answer is in 200 OK from the UAS > if (!has_body("application/sdp")) > return; > > if (isflagset(SRC_WS) && isbflagset(DST_WS)) > $var(rtpengine_flags) = "ICE=force-relay DTLS=passive"; > else if (isflagset(SRC_WS) && !isbflagset(DST_WS)) > $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; > else if (!isflagset(SRC_WS) && isbflagset(DST_WS)) > $var(rtpengine_flags) = "RTP/AVP replace-session-connection > replace-origin ICE=remove"; > else if (!isflagset(SRC_WS) && !isbflagset(DST_WS)) > $var(rtpengine_flags) = "RTP/AVP replace-session-connection > replace-origin ICE=remove"; > > # only enable transcoding if websocket call for now > if (isflagset(SRC_WS) || isbflagset(DST_WS)) { > rtpengine_answer("$var(rtpengine_flags)"); > } > } > > failure_route[missed_call] { > script_trace(1, "$rm from $si, rur=$ru", "me"); > xlog("[$ci/$T_branch_idx] failure_route:missed_call: incoming > failure response to $rm <- $T_reply_code/$T_ruri"); > if (t_was_cancelled()) { > xlog("[$ci/$T_branch_idx] was cancelled"); > exit; > } > do_accounting("db", "missed"); > > if (!t_relay()) { > send_reply("500","Internal Error"); > } > else { > xlog("[$ci/$T_branch_idx] Relay success $rm/$T_reply_code"); > } > } > > > > > > > > > > > Cheers, > Gerwin > > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Thu Jul 19 03:14:37 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 19 Jul 2018 10:14:37 +0300 Subject: [OpenSIPS-Users] mi_json returns invalid JSON In-Reply-To: References: Message-ID: <04329915-bc6d-0b21-547f-c114ccca3e66@opensips.org> Hi, Kirill! This doesn't seem to be an invalid JSON, but only a bad formatting of the date. I agree it should not include the "\n", because it is useless, but this doesn't make it invalid. Are you experiencing trouble with using JSON libraries that don't support this format? Best regards, Răzvan On 07/17/2018 12:58 PM, Kirill Galinurov wrote: > Hi. I use opensips mi_json module to get statistics. > But for some commands it returns incorrect JSON. > For example /json/dr_reload_status returns > > {"Partition": [{"value":"sipprovider_def", "attributes":{"Date": "Mon > Jul 16 07:10:58 2018 > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 07:11:59 2018 > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 07:11:59 2018 > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 07:12:54 2018 > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:12:54 2018 > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:13:51 2018 > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 07:13:52 2018 > "}}]} > > This is incorrect, because it shoulde not include "\n" symbols. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From Ben.Newlin at genesys.com Thu Jul 19 08:58:24 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 19 Jul 2018 12:58:24 +0000 Subject: [OpenSIPS-Users] mi_json returns invalid JSON In-Reply-To: <04329915-bc6d-0b21-547f-c114ccca3e66@opensips.org> References: <04329915-bc6d-0b21-547f-c114ccca3e66@opensips.org> Message-ID: I believe newline characters used within JSON must be escaped. So if the JSON actually contains "\n" that is invalid. It must be "\\n". Ben Newlin On 7/19/18, 3:19 AM, "Users on behalf of Răzvan Crainea" wrote: Hi, Kirill! This doesn't seem to be an invalid JSON, but only a bad formatting of the date. I agree it should not include the "\n", because it is useless, but this doesn't make it invalid. Are you experiencing trouble with using JSON libraries that don't support this format? Best regards, Răzvan On 07/17/2018 12:58 PM, Kirill Galinurov wrote: > Hi. I use opensips mi_json module to get statistics. > But for some commands it returns incorrect JSON. > For example /json/dr_reload_status returns > > {"Partition": [{"value":"sipprovider_def", "attributes":{"Date": "Mon > Jul 16 07:10:58 2018 > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 07:11:59 2018 > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 07:11:59 2018 > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 07:12:54 2018 > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:12:54 2018 > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:13:51 2018 > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 07:13:52 2018 > "}}]} > > This is incorrect, because it shoulde not include "\n" symbols. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From Ben.Newlin at genesys.com Thu Jul 19 09:03:09 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 19 Jul 2018 13:03:09 +0000 Subject: [OpenSIPS-Users] mi_json returns invalid JSON In-Reply-To: References: <04329915-bc6d-0b21-547f-c114ccca3e66@opensips.org> Message-ID: Actually, after further investigation I'm no longer sure that is true. According to http://www.json.org/ the "\n" is valid in JSON. Ben Newlin On 7/19/18, 8:58 AM, "Ben Newlin" wrote: I believe newline characters used within JSON must be escaped. So if the JSON actually contains "\n" that is invalid. It must be "\\n". Ben Newlin On 7/19/18, 3:19 AM, "Users on behalf of Răzvan Crainea" wrote: Hi, Kirill! This doesn't seem to be an invalid JSON, but only a bad formatting of the date. I agree it should not include the "\n", because it is useless, but this doesn't make it invalid. Are you experiencing trouble with using JSON libraries that don't support this format? Best regards, Răzvan On 07/17/2018 12:58 PM, Kirill Galinurov wrote: > Hi. I use opensips mi_json module to get statistics. > But for some commands it returns incorrect JSON. > For example /json/dr_reload_status returns > > {"Partition": [{"value":"sipprovider_def", "attributes":{"Date": "Mon > Jul 16 07:10:58 2018 > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 07:11:59 2018 > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 07:11:59 2018 > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 07:12:54 2018 > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:12:54 2018 > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:13:51 2018 > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 07:13:52 2018 > "}}]} > > This is incorrect, because it shoulde not include "\n" symbols. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From sastre.sebastian at gmail.com Thu Jul 19 20:03:14 2018 From: sastre.sebastian at gmail.com (Sebastian Sastre) Date: Thu, 19 Jul 2018 20:03:14 -0400 Subject: [OpenSIPS-Users] Proto WSS No Open TCP Connection after reinvite In-Reply-To: References: Message-ID: ​Razvan, Thanks ! I tried what you indicated but I don’t see the contact changing. Im taking care of the fix contacts where it needs to be bet but still on the bye it can’t find it. root at gcwregistrar151:~$ opensipsctl fifo dlg_list. *(Call Connected)* dialog:: ID=5820137817639 state:: 4 user_flags:: 0 timestart:: 1532043804 datestart:: 2018-07-19 19:43:24 timeout:: 1532044163 dateout:: 2018-07-19 19:49:23 callid:: fp436dll6pcmdqk78gn6 from_uri:: sip:user at domain.com to_uri:: sip:18889990000 at domain.com caller_tag:: 1i4vfmjico caller_contact:: sip:lccpphv2 at 192.168.202.3:51292;transport=wss;ob callee_cseq:: 0 caller_route_set:: caller_bind_addr:: wss:10.101.10.151:443 caller_sdp:: CALLEES:: callee:: callee_tag:: d651df12-c9c2-4db1-99ad-b15d6240ffee callee_contact:: sip:10.101.10.161:5060 caller_cseq:: 1094 callee_route_set:: callee_bind_addr:: udp:10.101.10.151:5060 callee_sdp:: root at gcwregistrar151:~$ opensipsctl fifo dlg_list *(Call on Hold )* dialog:: ID=5820137817639 state:: 4 user_flags:: 0 timestart:: 1532043804 datestart:: 2018-07-19 19:43:24 timeout:: 1532044163 dateout:: 2018-07-19 19:49:23 callid:: fp436dll6pcmdqk78gn6 from_uri:: sip:user at domain.com to_uri:: sip:18889990000 at domain.com caller_tag:: 1i4vfmjico caller_contact:: sip:lccpphv2 at 192.168.202.3:51292;transport=wss;ob callee_cseq:: 0 caller_route_set:: caller_bind_addr:: wss:10.101.10.151:443 caller_sdp:: CALLEES:: callee:: callee_tag:: d651df12-c9c2-4db1-99ad-b15d6240ffee callee_contact:: sip:10.101.10.161:5060 caller_cseq:: 1095 callee_route_set:: callee_bind_addr:: udp:10.101.10.151:5060 callee_sdp:: On Mon, Jul 16, 2018 at 7:55 AM, Răzvan Crainea wrote: > Hi, Sebastian! > > The re-invite probably generates a remote contact update. And if you don't > "fix" the contact on re-invites and their 200 OK, you might end up with > broken contacts in the dialog, thus sequential signaling will not work. > I suggest you do two things to debug this: > 1. remove the fix_route_dialog() call - the call should still be routed > according to RR information, presuming this information is correct. > 2. start the call, run `opensipsctl fifo dlg_list` and write down the > WSS's contact, then put the call on hold, and check again the contact. > > Best regards, > Răzvan > > > On 07/13/2018 09:19 PM, Sebastian Sastre wrote: > >> >> Hello, I’ve been experiencing a situation with Proto WSS. The scenario is >> very simple. A call is established from an Asterisk Box to Opensips (UDP) >> and finally a SipJs7.8 (WSS). Everything works great and we are able to >> register using mid registrar and pass calls thru. >> >> When an agent puts the call on hold a reinvite is correctly negotiated >> and the call is placed on hold and viceversa. However!, if the >> originating caller disconnects the call while still on hold, Asterisk will >> correctly terminate the dialog with a Bye but when OpenSIPs will complain >> about not finding a suitable tcp connection and responds with a 477 even >> after successfully matching and processing the dialog termination correctly. >> >> opensipsctl fifo list_tcp_conns shows the connection available. >> >> The only way I found of fixing this problem is by adding >> fix_route_dialog() on the sequential loose route. >> >> if (loose_route()) { >> if (is_method("BYE")) { >> if (!validate_dialog()){ >> fix_route_dialog(); >> } >> >> What do you guys think? >> Am I messing up something in the script or is this the correct way to >> address this problem? >> >> The funny thing is that there is no difference notable between the bye >> after hold and a regular bye without putting the call on hold. >> Here is the opensips log with the error and the trace. >> >> https://pastebin.com/BEJ6fAR8 >> >> Thanks ! >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From k.galinurov at gmail.com Fri Jul 20 03:29:13 2018 From: k.galinurov at gmail.com (Kirill Galinurov) Date: Fri, 20 Jul 2018 10:29:13 +0300 Subject: [OpenSIPS-Users] Users Digest, Vol 120, Issue 29 In-Reply-To: References: Message-ID: Yes it's wrong Data format. "Date":"Thu Jul 19 11:30:33 2018 2018-07-19 19:00 GMT+03:00 : > Send Users mailing list submissions to > users at lists.opensips.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request at lists.opensips.org > > You can reach the person managing the list at > users-owner at lists.opensips.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: mi_json returns invalid JSON (Răzvan Crainea) > 2. Re: mi_json returns invalid JSON (Ben Newlin) > 3. Re: mi_json returns invalid JSON (Ben Newlin) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 19 Jul 2018 10:14:37 +0300 > From: Răzvan Crainea > To: users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] mi_json returns invalid JSON > Message-ID: <04329915-bc6d-0b21-547f-c114ccca3e66 at opensips.org> > Content-Type: text/plain; charset=utf-8; format=flowed > > Hi, Kirill! > > This doesn't seem to be an invalid JSON, but only a bad formatting of > the date. I agree it should not include the "\n", because it is useless, > but this doesn't make it invalid. Are you experiencing trouble with > using JSON libraries that don't support this format? > > Best regards, > Răzvan > > On 07/17/2018 12:58 PM, Kirill Galinurov wrote: > > Hi. I use opensips mi_json module to get statistics. > > But for some commands it returns incorrect JSON. > > For example /json/dr_reload_status returns > > > > {"Partition": [{"value":"sipprovider_def", "attributes":{"Date": "Mon > > Jul 16 07:10:58 2018 > > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 07:11:59 2018 > > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 07:11:59 2018 > > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 07:12:54 2018 > > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:12:54 2018 > > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:13:51 2018 > > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 07:13:52 2018 > > "}}]} > > > > This is incorrect, because it shoulde not include "\n" symbols. > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > > > ------------------------------ > > Message: 2 > Date: Thu, 19 Jul 2018 12:58:24 +0000 > From: Ben Newlin > To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] mi_json returns invalid JSON > Message-ID: > Content-Type: text/plain; charset=UTF-8 > > I believe newline characters used within JSON must be escaped. So if the > JSON actually contains "\n" that is invalid. It must be "\\n". > > Ben Newlin > > On 7/19/18, 3:19 AM, "Users on behalf of Răzvan Crainea" < > users-bounces at lists.opensips.org on behalf of razvan at opensips.org> wrote: > > Hi, Kirill! > > This doesn't seem to be an invalid JSON, but only a bad formatting of > the date. I agree it should not include the "\n", because it is > useless, > but this doesn't make it invalid. Are you experiencing trouble with > using JSON libraries that don't support this format? > > Best regards, > Răzvan > > On 07/17/2018 12:58 PM, Kirill Galinurov wrote: > > Hi. I use opensips mi_json module to get statistics. > > But for some commands it returns incorrect JSON. > > For example /json/dr_reload_status returns > > > > {"Partition": [{"value":"sipprovider_def", "attributes":{"Date": > "Mon > > Jul 16 07:10:58 2018 > > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 07:11:59 > 2018 > > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 07:11:59 > 2018 > > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 07:12:54 > 2018 > > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:12:54 2018 > > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:13:51 2018 > > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 07:13:52 > 2018 > > "}}]} > > > > This is incorrect, because it shoulde not include "\n" symbols. > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ------------------------------ > > Message: 3 > Date: Thu, 19 Jul 2018 13:03:09 +0000 > From: Ben Newlin > To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] mi_json returns invalid JSON > Message-ID: > Content-Type: text/plain; charset=UTF-8 > > Actually, after further investigation I'm no longer sure that is true. > According to http://www.json.org/ the "\n" is valid in JSON. > > Ben Newlin > > On 7/19/18, 8:58 AM, "Ben Newlin" wrote: > > I believe newline characters used within JSON must be escaped. So if > the JSON actually contains "\n" that is invalid. It must be "\\n". > > Ben Newlin > > On 7/19/18, 3:19 AM, "Users on behalf of Răzvan Crainea" < > users-bounces at lists.opensips.org on behalf of razvan at opensips.org> wrote: > > Hi, Kirill! > > This doesn't seem to be an invalid JSON, but only a bad formatting > of > the date. I agree it should not include the "\n", because it is > useless, > but this doesn't make it invalid. Are you experiencing trouble > with > using JSON libraries that don't support this format? > > Best regards, > Răzvan > > On 07/17/2018 12:58 PM, Kirill Galinurov wrote: > > Hi. I use opensips mi_json module to get statistics. > > But for some commands it returns incorrect JSON. > > For example /json/dr_reload_status returns > > > > {"Partition": [{"value":"sipprovider_def", "attributes":{"Date": > "Mon > > Jul 16 07:10:58 2018 > > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 > 07:11:59 2018 > > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 > 07:11:59 2018 > > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 > 07:12:54 2018 > > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:12:54 > 2018 > > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:13:51 > 2018 > > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 > 07:13:52 2018 > > "}}]} > > > > This is incorrect, because it shoulde not include "\n" symbols. > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ------------------------------ > > End of Users Digest, Vol 120, Issue 29 > ************************************** > -------------- next part -------------- An HTML attachment was scrubbed... URL: From sastre.sebastian at gmail.com Mon Jul 23 11:43:00 2018 From: sastre.sebastian at gmail.com (Sebastian Sastre) Date: Mon, 23 Jul 2018 11:43:00 -0400 Subject: [OpenSIPS-Users] Proto WSS No Open TCP Connection after reinvite In-Reply-To: References: Message-ID: Hey Razvan, I’ve been playing around a lot with this but I can’t seem to make it work. Whatever I do without the fix route doesn’t find a suitable tcp connection. Do you see something on the dlg_list before that would indicate the problem Or is there any other debug I can use ? Thanks again ! On Thu, Jul 19, 2018 at 8:03 PM, Sebastian Sastre < sastre.sebastian at gmail.com> wrote: > ​Razvan, > Thanks ! I tried what you indicated but I don’t see the contact changing. > Im taking care of the fix contacts where it needs to be bet but still on > the bye it can’t find it. > > > root at gcwregistrar151:~$ opensipsctl fifo dlg_list. *(Call Connected)* > dialog:: ID=5820137817639 > state:: 4 > user_flags:: 0 > timestart:: 1532043804 > datestart:: 2018-07-19 19:43:24 > timeout:: 1532044163 > dateout:: 2018-07-19 19:49:23 > callid:: fp436dll6pcmdqk78gn6 > from_uri:: sip:user at domain.com > to_uri:: sip:18889990000 at domain.com > caller_tag:: 1i4vfmjico > caller_contact:: sip:lccpphv2 at 192.168.202.3:51292;transport=wss;ob > callee_cseq:: 0 > caller_route_set:: > caller_bind_addr:: wss:10.101.10.151:443 > caller_sdp:: > CALLEES:: > callee:: > callee_tag:: d651df12-c9c2-4db1-99ad-b15d6240ffee > callee_contact:: sip:10.101.10.161:5060 > caller_cseq:: 1094 > callee_route_set:: > callee_bind_addr:: udp:10.101.10.151:5060 > callee_sdp:: > > root at gcwregistrar151:~$ opensipsctl fifo dlg_list *(Call on Hold )* > dialog:: ID=5820137817639 > state:: 4 > user_flags:: 0 > timestart:: 1532043804 > datestart:: 2018-07-19 19:43:24 > timeout:: 1532044163 > dateout:: 2018-07-19 19:49:23 > callid:: fp436dll6pcmdqk78gn6 > from_uri:: sip:user at domain.com > to_uri:: sip:18889990000 at domain.com > caller_tag:: 1i4vfmjico > caller_contact:: sip:lccpphv2 at 192.168.202.3:51292;transport=wss;ob > callee_cseq:: 0 > caller_route_set:: > caller_bind_addr:: wss:10.101.10.151:443 > caller_sdp:: > CALLEES:: > callee:: > callee_tag:: d651df12-c9c2-4db1-99ad-b15d6240ffee > callee_contact:: sip:10.101.10.161:5060 > caller_cseq:: 1095 > callee_route_set:: > callee_bind_addr:: udp:10.101.10.151:5060 > callee_sdp:: > > > On Mon, Jul 16, 2018 at 7:55 AM, Răzvan Crainea > wrote: > >> Hi, Sebastian! >> >> The re-invite probably generates a remote contact update. And if you >> don't "fix" the contact on re-invites and their 200 OK, you might end up >> with broken contacts in the dialog, thus sequential signaling will not work. >> I suggest you do two things to debug this: >> 1. remove the fix_route_dialog() call - the call should still be routed >> according to RR information, presuming this information is correct. >> 2. start the call, run `opensipsctl fifo dlg_list` and write down the >> WSS's contact, then put the call on hold, and check again the contact. >> >> Best regards, >> Răzvan >> >> >> On 07/13/2018 09:19 PM, Sebastian Sastre wrote: >> >>> >>> Hello, I’ve been experiencing a situation with Proto WSS. The scenario >>> is very simple. A call is established from an Asterisk Box to Opensips >>> (UDP) and finally a SipJs7.8 (WSS). Everything works great and we are able >>> to register using mid registrar and pass calls thru. >>> >>> When an agent puts the call on hold a reinvite is correctly negotiated >>> and the call is placed on hold and viceversa. However!, if the >>> originating caller disconnects the call while still on hold, Asterisk will >>> correctly terminate the dialog with a Bye but when OpenSIPs will complain >>> about not finding a suitable tcp connection and responds with a 477 even >>> after successfully matching and processing the dialog termination correctly. >>> >>> opensipsctl fifo list_tcp_conns shows the connection available. >>> >>> The only way I found of fixing this problem is by adding >>> fix_route_dialog() on the sequential loose route. >>> >>> if (loose_route()) { >>> if (is_method("BYE")) { >>> if (!validate_dialog()){ >>> fix_route_dialog(); >>> } >>> >>> What do you guys think? >>> Am I messing up something in the script or is this the correct way to >>> address this problem? >>> >>> The funny thing is that there is no difference notable between the bye >>> after hold and a regular bye without putting the call on hold. >>> Here is the opensips log with the error and the trace. >>> >>> https://pastebin.com/BEJ6fAR8 >>> >>> Thanks ! >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> -- >> Răzvan Crainea >> OpenSIPS Core Developer >> http://www.opensips-solutions.com >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Jul 23 11:47:37 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 23 Jul 2018 18:47:37 +0300 Subject: [OpenSIPS-Users] Proto WSS No Open TCP Connection after reinvite In-Reply-To: References: Message-ID: <4a54c190-d8ad-24d6-e2a1-de88fdc18435@opensips.org> Hi, Sebastien! It looks like the contact is not changing. Can you indicate what connection he is trying to find, i.e. what IP, port and proto? You should see them in the error line. Best regards, Răzvan On 07/23/2018 06:43 PM, Sebastian Sastre wrote: > Hey Razvan, > > I’ve been playing around a lot with this but I can’t seem to make it > work. Whatever I do without the fix route doesn’t find a suitable tcp > connection. > > Do you see something on the dlg_list before that would indicate the > problem Or is there any other debug I can use ? > > Thanks again ! > > > On Thu, Jul 19, 2018 at 8:03 PM, Sebastian Sastre > > wrote: > > ​Razvan, > Thanks ! I tried what you indicated but I don’t see the contact > changing. Im taking care of the fix contacts where it needs to be > bet but still on the bye it can’t find it. > > > root at gcwregistrar151:~$ opensipsctl fifo dlg_list. *(Call Connected)* > dialog::  ID=5820137817639 >         state:: 4 >         user_flags:: 0 >         timestart:: 1532043804 >         datestart:: 2018-07-19 19:43:24 >         timeout:: 1532044163 >         dateout:: 2018-07-19 19:49:23 >         callid:: fp436dll6pcmdqk78gn6 >         from_uri:: sip:user at domain.com >         to_uri:: sip:18889990000 at domain.com > >         caller_tag:: 1i4vfmjico >         caller_contact:: sip:lccpphv2 at 192.168.202.3 > :51292;transport=wss;ob >         callee_cseq:: 0 >         caller_route_set:: >         caller_bind_addr:: wss:10.101.10.151:443 > >         caller_sdp:: >         CALLEES:: >                 callee:: >                         callee_tag:: > d651df12-c9c2-4db1-99ad-b15d6240ffee >                         callee_contact:: sip:10.101.10.161:5060 > >                         caller_cseq:: 1094 >                         callee_route_set:: >                         callee_bind_addr:: udp:10.101.10.151:5060 > >                         callee_sdp:: > > root at gcwregistrar151:~$ opensipsctl fifo dlg_list *(Call on Hold )* > dialog::  ID=5820137817639 >         state:: 4 >         user_flags:: 0 >         timestart:: 1532043804 >         datestart:: 2018-07-19 19:43:24 >         timeout:: 1532044163 >         dateout:: 2018-07-19 19:49:23 >         callid:: fp436dll6pcmdqk78gn6 >         from_uri:: sip:user at domain.com >         to_uri:: sip:18889990000 at domain.com > >         caller_tag:: 1i4vfmjico >         caller_contact:: sip:lccpphv2 at 192.168.202.3 > :51292;transport=wss;ob >         callee_cseq:: 0 >         caller_route_set:: >         caller_bind_addr:: wss:10.101.10.151:443 > >         caller_sdp:: >         CALLEES:: >                 callee:: >                         callee_tag:: > d651df12-c9c2-4db1-99ad-b15d6240ffee >                         callee_contact:: sip:10.101.10.161:5060 > >                         caller_cseq:: 1095 >                         callee_route_set:: >                         callee_bind_addr:: udp:10.101.10.151:5060 > >                         callee_sdp:: > > > On Mon, Jul 16, 2018 at 7:55 AM, Răzvan Crainea > wrote: > > Hi, Sebastian! > > The re-invite probably generates a remote contact update. And if > you don't "fix" the contact on re-invites and their 200 OK, you > might end up with broken contacts in the dialog, thus sequential > signaling will not work. > I suggest you do two things to debug this: > 1. remove the fix_route_dialog() call - the call should still be > routed according to RR information, presuming this information > is correct. > 2. start the call, run `opensipsctl fifo dlg_list` and write > down the WSS's contact, then put the call on hold, and check > again the contact. > > Best regards, > Răzvan > > > On 07/13/2018 09:19 PM, Sebastian Sastre wrote: > > > Hello, I’ve been experiencing a situation with Proto WSS. > The scenario is very simple. A call is established from an > Asterisk Box to Opensips (UDP) and finally a SipJs7.8 (WSS). > Everything works great and we are able to register using mid > registrar and pass calls thru. > > When an agent puts the call on hold a reinvite is correctly > negotiated and the call is placed on hold and viceversa. > However!, if the originating caller disconnects the call > while still on hold, Asterisk will correctly terminate the > dialog with a Bye but when OpenSIPs will complain about not > finding a suitable tcp connection and responds with a 477 > even after successfully matching and processing the dialog > termination correctly. > > opensipsctl fifo list_tcp_conns  shows the connection available. > > The only way I found of fixing this problem is by adding > fix_route_dialog() on the sequential loose route. > > if (loose_route()) { > if (is_method("BYE")) { >                          if (!validate_dialog()){ >                                fix_route_dialog(); >                          } > > What do you guys think? > Am I messing up something in the script or is this the > correct way to address this problem? > > The funny thing is that there is no difference notable > between the bye after hold and a regular bye without putting > the call on hold. > Here is the opensips log with the error and the trace. > > https://pastebin.com/BEJ6fAR8 > > Thanks ! > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From sastre.sebastian at gmail.com Mon Jul 23 11:58:02 2018 From: sastre.sebastian at gmail.com (Sebastian Sastre) Date: Mon, 23 Jul 2018 11:58:02 -0400 Subject: [OpenSIPS-Users] Proto WSS No Open TCP Connection after reinvite In-Reply-To: <4a54c190-d8ad-24d6-e2a1-de88fdc18435@opensips.org> References: <4a54c190-d8ad-24d6-e2a1-de88fdc18435@opensips.org> Message-ID: I put a full debug on this paste https://pastebin.com/BEJ6fAR8 Should be Jul 13 11:42:39 gcwregistrar151 /sbin/opensips[3647]: ERROR:tm:msg_send: send() to 192.0.2.246:443 for proto wss/6 failed Thanks On Mon, Jul 23, 2018 at 11:47 AM, Răzvan Crainea wrote: > Hi, Sebastien! > > It looks like the contact is not changing. Can you indicate what > connection he is trying to find, i.e. what IP, port and proto? You should > see them in the error line. > > Best regards, > Răzvan > > On 07/23/2018 06:43 PM, Sebastian Sastre wrote: > >> Hey Razvan, >> >> I’ve been playing around a lot with this but I can’t seem to make it >> work. Whatever I do without the fix route doesn’t find a suitable tcp >> connection. >> >> Do you see something on the dlg_list before that would indicate the >> problem Or is there any other debug I can use ? >> >> Thanks again ! >> >> >> On Thu, Jul 19, 2018 at 8:03 PM, Sebastian Sastre < >> sastre.sebastian at gmail.com > wrote: >> >> ​Razvan, >> Thanks ! I tried what you indicated but I don’t see the contact >> changing. Im taking care of the fix contacts where it needs to be >> bet but still on the bye it can’t find it. >> >> >> root at gcwregistrar151:~$ opensipsctl fifo dlg_list. *(Call Connected)* >> dialog:: ID=5820137817639 >> state:: 4 >> user_flags:: 0 >> timestart:: 1532043804 >> datestart:: 2018-07-19 19:43:24 >> timeout:: 1532044163 >> dateout:: 2018-07-19 19:49:23 >> callid:: fp436dll6pcmdqk78gn6 >> from_uri:: sip:user at domain.com > > >> to_uri:: sip:18889990000 at domain.com >> >> caller_tag:: 1i4vfmjico >> caller_contact:: sip:lccpphv2 at 192.168.202.3 >> :51292;transport=wss;ob >> callee_cseq:: 0 >> caller_route_set:: >> caller_bind_addr:: wss:10.101.10.151:443 >> >> caller_sdp:: >> CALLEES:: >> callee:: >> callee_tag:: >> d651df12-c9c2-4db1-99ad-b15d6240ffee >> callee_contact:: sip:10.101.10.161:5060 >> >> caller_cseq:: 1094 >> callee_route_set:: >> callee_bind_addr:: udp:10.101.10.151:5060 >> >> callee_sdp:: >> >> root at gcwregistrar151:~$ opensipsctl fifo dlg_list *(Call on Hold )* >> dialog:: ID=5820137817639 >> state:: 4 >> user_flags:: 0 >> timestart:: 1532043804 >> datestart:: 2018-07-19 19:43:24 >> timeout:: 1532044163 >> dateout:: 2018-07-19 19:49:23 >> callid:: fp436dll6pcmdqk78gn6 >> from_uri:: sip:user at domain.com > > >> to_uri:: sip:18889990000 at domain.com >> >> caller_tag:: 1i4vfmjico >> caller_contact:: sip:lccpphv2 at 192.168.202.3 >> :51292;transport=wss;ob >> callee_cseq:: 0 >> caller_route_set:: >> caller_bind_addr:: wss:10.101.10.151:443 >> >> caller_sdp:: >> CALLEES:: >> callee:: >> callee_tag:: >> d651df12-c9c2-4db1-99ad-b15d6240ffee >> callee_contact:: sip:10.101.10.161:5060 >> >> caller_cseq:: 1095 >> callee_route_set:: >> callee_bind_addr:: udp:10.101.10.151:5060 >> >> callee_sdp:: >> >> >> On Mon, Jul 16, 2018 at 7:55 AM, Răzvan Crainea > > wrote: >> >> Hi, Sebastian! >> >> The re-invite probably generates a remote contact update. And if >> you don't "fix" the contact on re-invites and their 200 OK, you >> might end up with broken contacts in the dialog, thus sequential >> signaling will not work. >> I suggest you do two things to debug this: >> 1. remove the fix_route_dialog() call - the call should still be >> routed according to RR information, presuming this information >> is correct. >> 2. start the call, run `opensipsctl fifo dlg_list` and write >> down the WSS's contact, then put the call on hold, and check >> again the contact. >> >> Best regards, >> Răzvan >> >> >> On 07/13/2018 09:19 PM, Sebastian Sastre wrote: >> >> >> Hello, I’ve been experiencing a situation with Proto WSS. >> The scenario is very simple. A call is established from an >> Asterisk Box to Opensips (UDP) and finally a SipJs7.8 (WSS). >> Everything works great and we are able to register using mid >> registrar and pass calls thru. >> >> When an agent puts the call on hold a reinvite is correctly >> negotiated and the call is placed on hold and viceversa. >> However!, if the originating caller disconnects the call >> while still on hold, Asterisk will correctly terminate the >> dialog with a Bye but when OpenSIPs will complain about not >> finding a suitable tcp connection and responds with a 477 >> even after successfully matching and processing the dialog >> termination correctly. >> >> opensipsctl fifo list_tcp_conns shows the connection >> available. >> >> The only way I found of fixing this problem is by adding >> fix_route_dialog() on the sequential loose route. >> >> if (loose_route()) { >> if (is_method("BYE")) { >> if (!validate_dialog()){ >> fix_route_dialog(); >> } >> >> What do you guys think? >> Am I messing up something in the script or is this the >> correct way to address this problem? >> >> The funny thing is that there is no difference notable >> between the bye after hold and a regular bye without putting >> the call on hold. >> Here is the opensips log with the error and the trace. >> >> https://pastebin.com/BEJ6fAR8 >> >> Thanks ! >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- Răzvan Crainea >> OpenSIPS Core Developer >> http://www.opensips-solutions.com >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Mon Jul 23 22:33:24 2018 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Mon, 23 Jul 2018 23:33:24 -0300 Subject: [OpenSIPS-Users] mid_registrar cluster Message-ID: <1532399604.2081.0@smtp.networklab.ca> Hello Everyone, I was wonder if mid registrar will support g flag for global lookup if AOR no belong to local cluster node Also when I set mode 2 Contact header is not formated correctly. config #### USeR Location module loadmodule "usrloc.so" modparam("usrloc", "db_url", "") modparam("usrloc", "use_domain", 1) modparam("usrloc", "working_mode_preset", "federation-cachedb-cluster") modparam("usrloc", "location_cluster", 1) modparam("usrloc", "cachedb_url", "mongodb://") modparam("usrloc", "restart_persistency", "sync-from-cluster") modparam("usrloc","nat_bflag","NATED_CALLEE") #### REGISTRAR module loadmodule "mid_registrar.so" modparam("mid_registrar", "mode", 2) modparam("mid_registrar", "received_avp", "$avp(RECEIVED)") modparam("mid_registrar", "max_contacts", 10) modparam("mid_registrar", "tcp_persistent_flag", "TCP_PERSIST_REGISTRATIONS") modparam("mid_registrar", "outgoing_expires", 7200) modparam("mid_registrar", "contact_id_param", "ctid") log Jul 23 21:26:06 aitossbc01 /usr/sbin/opensips[28045]: ERROR:core:parse_sip_msg_uri: bad uri Jul 23 21:26:06 aitossbc01 /usr/sbin/opensips[28045]: ERROR:core:pv_get_ruri: failed to parse the R-URI Jul 23 21:26:06 aitossbc01 /usr/sbin/opensips[28045]: looking up [] Jul 23 21:26:06 aitossbc01 /usr/sbin/opensips[28045]: WARNING:mid_registrar:mid_reg_lookup: unsupported flag g volga629 From sanjeevt510 at yahoo.com Mon Jul 23 22:51:40 2018 From: sanjeevt510 at yahoo.com (Sanjeev Sharma) Date: Tue, 24 Jul 2018 02:51:40 +0000 (UTC) Subject: [OpenSIPS-Users] Media Issue once the UA - connected References: <258243532.1172160.1532400700468.ref@mail.yahoo.com> Message-ID: <258243532.1172160.1532400700468@mail.yahoo.com> Hi , i am  new to opensips world and i installed the openSIPS version 2.2 , facing issue of media. Setup is my configuration is like 1) UAC request come through fortigate firewall on public address (1234 at 49.121.121.121)  and then passed to my opensip machine (centos 7 , opensip version 2.2) having local address ( natted with public address) 2) registration of UAC is fine and if UAC are on same network lets say one client on my laptop and another on mobile device (both on same ISP WIFI) then voice is going through between the UAC but i switch the network of the mobile to telco service provider then their is no media pass on in between the UA or some time only one UA able to listen the other side. i looked online and search lots of stuffs related to this and changed in configuration but unable to solve or find what and where i am getting. i tried TCP dump for both kinds of call i.e having voice or no voice but unable to identify the difference between calls having voice and no voice. Since i am new to this setup and configuration , just stuck due to above problem . Please suggest where to look and what could be the possible reason. Currently all traffic ( i.e all ports open) being allow from firewall to machine  ThanksSanjeev  -------------- next part -------------- An HTML attachment was scrubbed... URL: From mimmo at ats.it Tue Jul 24 05:20:41 2018 From: mimmo at ats.it (Simone Sestini) Date: Tue, 24 Jul 2018 11:20:41 +0200 Subject: [OpenSIPS-Users] Mediaproxy and Debian 8.11 In-Reply-To: References: Message-ID: Dear list.. I'm trying to install mediaproxy software on Debian 8.11. I'm looking around for a solution.. because i'm receiving test-rtp:/# apt-get install mediaproxy-common  mediaproxy-relay mediaproxy-web-sessions Lettura elenco dei pacchetti... Fatto Generazione albero delle dipendenze Lettura informazioni sullo stato... Fatto Alcuni pacchetti non possono essere installati. Questo pu√≤ voler dire che √® stata richiesta una situazione impossibile oppure, se si sta usando una distribuzione in sviluppo, che alcuni pacchetti richiesti non sono ancora stati creati o sono stati rimossi da Incoming. Le seguenti informazioni possono aiutare a risolvere la situazione: I seguenti pacchetti hanno dipendenze non soddisfatte: * mediaproxy-common : Dipende: ***libiptc0* ma non √® installabile* E: Impossibile correggere i problemi, ci sono pacchetti danneggiati bloccati. test-rtp:/# Praticaly mediaproxy-common depend of *libiptc0 *but it's not found on debian. The only lib exist are lrwxrwxrwx 1 root root   16 nov  8  2014 /lib/libiptc.so -> libiptc.so.0.0.0 lrwxrwxrwx 1 root root   16 nov  8  2014 /lib/libiptc.so.0 -> libiptc.so.0.0.0 -rw-r--r-- 1 root root 5816 nov  8  2014 /lib/libiptc.so.0.0.0 and i tried to add libiptc0 links too but it dosn't fix.. lrwxrwxrwx 1 root root   16 lug 23 12:49 /lib/libiptc0.so -> libiptc.so.0.0.0 lrwxrwxrwx 1 root root   16 lug 23 12:50 /lib/libiptc0.so.0 -> libiptc.so.0.0.0 Any idea ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jul 24 06:11:05 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 24 Jul 2018 13:11:05 +0300 Subject: [OpenSIPS-Users] Media Issue once the UA - connected In-Reply-To: <258243532.1172160.1532400700468@mail.yahoo.com> References: <258243532.1172160.1532400700468.ref@mail.yahoo.com> <258243532.1172160.1532400700468@mail.yahoo.com> Message-ID: Hi Sanjeev, As I understand correctly, you end up connecting into your opensips devices from different networks - a devices from the private network (same as opensips) and another device from the public network. But note that bidirectional direct communication between the 2 devices is not possible, as the public device cannot send traffic to a private IP/destination. Depending on the opensips cfg, the SIP signaling works, as OpenSIPS will act as a bridge between the 2 networks. But this is not true for RTP, as RTP goes directly between the 2 devices. So, what you need is to use a media relay acting a bridge between the 2 networks. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 07/24/2018 05:51 AM, Sanjeev Sharma via Users wrote: > Hi , > > i am new to opensips world and i installed the openSIPS version 2.2 , > facing issue of media. Setup is my configuration is like > 1) UAC request come through fortigate firewall on public address > (1234 at 49.121.121.121) and then passed to my opensip machine (centos 7 > , opensip version 2.2) having local address ( natted with public address) > 2) registration of UAC is fine and if UAC are on same network lets say > one client on my laptop and another on mobile device (both on same ISP > WIFI) then voice is going through between the UAC but i switch the > network of the mobile to telco service provider then their is no media > pass on in between the UA or some time only one UA able to listen the > other side. > > i looked online and search lots of stuffs related to this and changed > in configuration but unable to solve or find what and where i am > getting. i tried TCP dump for both kinds of call i.e having voice or > no voice but unable to identify the difference between calls having > voice and no voice. > > Since i am new to this setup and configuration , just stuck due to > above problem . > > Please suggest where to look and what could be the possible reason. > Currently all traffic ( i.e all ports open) being allow from firewall > to machine > > > Thanks > Sanjeev > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From dan at ag-projects.com Tue Jul 24 06:32:42 2018 From: dan at ag-projects.com (Dan Pascu) Date: Tue, 24 Jul 2018 13:32:42 +0300 Subject: [OpenSIPS-Users] Mediaproxy and Debian 8.11 In-Reply-To: References: Message-ID: <75F34A6B-EA04-4BEB-B905-8B29E061B50A@ag-projects.com> Please paste apt's sources list (you can delete private repositories if you have any, I only want to see debian's and ag-projects' repositories). Alternatively you can paste the output from apt-cache policy iptables mediaproxy-common On 24 Jul 2018, at 12:20, Simone Sestini wrote: > Dear list.. > > I'm trying to install mediaproxy software on Debian 8.11. > > I'm looking around for a solution.. because i'm receiving > test-rtp:/# apt-get install mediaproxy-common mediaproxy-relay mediaproxy-web-sessions > Lettura elenco dei pacchetti... Fatto > Generazione albero delle dipendenze > Lettura informazioni sullo stato... Fatto > Alcuni pacchetti non possono essere installati. Questo pu√≤ voler dire > che √® stata richiesta una situazione impossibile oppure, se si sta > usando una distribuzione in sviluppo, che alcuni pacchetti richiesti > non sono ancora stati creati o sono stati rimossi da Incoming. > Le seguenti informazioni possono aiutare a risolvere la situazione: > > I seguenti pacchetti hanno dipendenze non soddisfatte: > mediaproxy-common : Dipende: libiptc0 ma non √® installabile > E: Impossibile correggere i problemi, ci sono pacchetti danneggiati bloccati. > test-rtp:/# > > Praticaly mediaproxy-common depend of libiptc0 but it's not found on debian. > > The only lib exist are > lrwxrwxrwx 1 root root 16 nov 8 2014 /lib/libiptc.so -> libiptc.so.0.0.0 > lrwxrwxrwx 1 root root 16 nov 8 2014 /lib/libiptc.so.0 -> libiptc.so.0.0.0 > -rw-r--r-- 1 root root 5816 nov 8 2014 /lib/libiptc.so.0.0.0 > > and i tried to add libiptc0 links too but it dosn't fix.. > > lrwxrwxrwx 1 root root 16 lug 23 12:49 /lib/libiptc0.so -> libiptc.so.0.0.0 > lrwxrwxrwx 1 root root 16 lug 23 12:50 /lib/libiptc0.so.0 -> libiptc.so.0.0.0 > > Any idea ? > > Regards > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dan From mimmo at ats.it Tue Jul 24 06:48:04 2018 From: mimmo at ats.it (Simone Sestini) Date: Tue, 24 Jul 2018 12:48:04 +0200 Subject: [OpenSIPS-Users] Mediaproxy and Debian 8.11 In-Reply-To: <75F34A6B-EA04-4BEB-B905-8B29E061B50A@ag-projects.com> References: <75F34A6B-EA04-4BEB-B905-8B29E061B50A@ag-projects.com> Message-ID: <3d98da49-f7b0-d3a7-65ae-52ddabf7974c@ats.it> Hi, only the standard and official repository here.. ------- deb http://ftp.it.debian.org/debian/ jessie main deb-src http://ftp.it.debian.org/debian/ jessie main deb http://security.debian.org/ jessie/updates main deb-src http://security.debian.org/ jessie/updates main # jessie-updates, previously known as 'volatile' deb http://ftp.it.debian.org/debian/ jessie-updates main deb-src http://ftp.it.debian.org/debian/ jessie-updates main # AG Projects software deb http://ag-projects.com/debian unstable main deb-src http://ag-projects.com/debian unstable main ------- and here the output for the  policy you requested. test-rtp:~# apt-cache policy iptables mediaproxy-common iptables:   Installato: 1.4.21-2+b1   Candidato:  1.4.21-2+b1   Tabella versione:  *** 1.4.21-2+b1 0         500 http://ftp.it.debian.org/debian/ jessie/main amd64 Packages         100 /var/lib/dpkg/status mediaproxy-common:   Installato: (nessuno)   Candidato:  2.6.6   Tabella versione:      2.6.6 0         500 http://ag-projects.com/debian/ unstable/main amd64 Packages Thanks Il 24/07/18 12:32, Dan Pascu ha scritto: > Please paste apt's sources list (you can delete private repositories if you have any, I only want to see debian's and ag-projects' repositories). Alternatively you can paste the output from > > apt-cache policy iptables mediaproxy-common > > On 24 Jul 2018, at 12:20, Simone Sestini wrote: > >> Dear list.. >> >> I'm trying to install mediaproxy software on Debian 8.11. >> >> I'm looking around for a solution.. because i'm receiving >> test-rtp:/# apt-get install mediaproxy-common mediaproxy-relay mediaproxy-web-sessions >> Lettura elenco dei pacchetti... Fatto >> Generazione albero delle dipendenze >> Lettura informazioni sullo stato... Fatto >> Alcuni pacchetti non possono essere installati. Questo pu√≤ voler dire >> che √® stata richiesta una situazione impossibile oppure, se si sta >> usando una distribuzione in sviluppo, che alcuni pacchetti richiesti >> non sono ancora stati creati o sono stati rimossi da Incoming. >> Le seguenti informazioni possono aiutare a risolvere la situazione: >> >> I seguenti pacchetti hanno dipendenze non soddisfatte: >> mediaproxy-common : Dipende: libiptc0 ma non √® installabile >> E: Impossibile correggere i problemi, ci sono pacchetti danneggiati bloccati. >> test-rtp:/# >> >> Praticaly mediaproxy-common depend of libiptc0 but it's not found on debian. >> >> The only lib exist are >> lrwxrwxrwx 1 root root 16 nov 8 2014 /lib/libiptc.so -> libiptc.so.0.0.0 >> lrwxrwxrwx 1 root root 16 nov 8 2014 /lib/libiptc.so.0 -> libiptc.so.0.0.0 >> -rw-r--r-- 1 root root 5816 nov 8 2014 /lib/libiptc.so.0.0.0 >> >> and i tried to add libiptc0 links too but it dosn't fix.. >> >> lrwxrwxrwx 1 root root 16 lug 23 12:49 /lib/libiptc0.so -> libiptc.so.0.0.0 >> lrwxrwxrwx 1 root root 16 lug 23 12:50 /lib/libiptc0.so.0 -> libiptc.so.0.0.0 >> >> Any idea ? >> >> Regards >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > Dan > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From dan at ag-projects.com Tue Jul 24 07:27:03 2018 From: dan at ag-projects.com (Dan Pascu) Date: Tue, 24 Jul 2018 14:27:03 +0300 Subject: [OpenSIPS-Users] Mediaproxy and Debian 8.11 In-Reply-To: <3d98da49-f7b0-d3a7-65ae-52ddabf7974c@ats.it> References: <75F34A6B-EA04-4BEB-B905-8B29E061B50A@ag-projects.com> <3d98da49-f7b0-d3a7-65ae-52ddabf7974c@ats.it> Message-ID: Your problem is that you mix debian 8.11 (jessie) with ag-projects' debian unstable repositories. Change your ag-projects repositories to read 'jessie main' instead of 'unstable main', rerun apt-get update and it should work. You might also consider adding the jessie-backports repository: deb http://ftp.it.debian.org/debian/ jessie-bakcports main deb-src http://ftp.it.debian.org/debian/ jessie-bakcports main Some of our software needs newer versions of some packages that are only available in the jessie-backports repository (not sure if mediaproxy is one of them, but others like blink or sipsimple are). On 24 Jul 2018, at 13:48, Simone Sestini wrote: > Hi, > > only the standard and official repository here.. > > ------- > deb http://ftp.it.debian.org/debian/ jessie main > deb-src http://ftp.it.debian.org/debian/ jessie main > > deb http://security.debian.org/ jessie/updates main > deb-src http://security.debian.org/ jessie/updates main > > # jessie-updates, previously known as 'volatile' > deb http://ftp.it.debian.org/debian/ jessie-updates main > deb-src http://ftp.it.debian.org/debian/ jessie-updates main > > # AG Projects software > deb http://ag-projects.com/debian unstable main > deb-src http://ag-projects.com/debian unstable main > > ------- > > and here the output for the policy you requested. > > test-rtp:~# apt-cache policy iptables mediaproxy-common > iptables: > Installato: 1.4.21-2+b1 > Candidato: 1.4.21-2+b1 > Tabella versione: > *** 1.4.21-2+b1 0 > 500 http://ftp.it.debian.org/debian/ jessie/main amd64 Packages > 100 /var/lib/dpkg/status > mediaproxy-common: > Installato: (nessuno) > Candidato: 2.6.6 > Tabella versione: > 2.6.6 0 > 500 http://ag-projects.com/debian/ unstable/main amd64 Packages > > > Thanks > > > Il 24/07/18 12:32, Dan Pascu ha scritto: >> Please paste apt's sources list (you can delete private repositories if you have any, I only want to see debian's and ag-projects' repositories). Alternatively you can paste the output from >> >> apt-cache policy iptables mediaproxy-common >> >> On 24 Jul 2018, at 12:20, Simone Sestini wrote: >> >>> Dear list.. >>> >>> I'm trying to install mediaproxy software on Debian 8.11. >>> >>> I'm looking around for a solution.. because i'm receiving >>> test-rtp:/# apt-get install mediaproxy-common mediaproxy-relay mediaproxy-web-sessions >>> Lettura elenco dei pacchetti... Fatto >>> Generazione albero delle dipendenze >>> Lettura informazioni sullo stato... Fatto >>> Alcuni pacchetti non possono essere installati. Questo pu√≤ voler dire >>> che √® stata richiesta una situazione impossibile oppure, se si sta >>> usando una distribuzione in sviluppo, che alcuni pacchetti richiesti >>> non sono ancora stati creati o sono stati rimossi da Incoming. >>> Le seguenti informazioni possono aiutare a risolvere la situazione: >>> >>> I seguenti pacchetti hanno dipendenze non soddisfatte: >>> mediaproxy-common : Dipende: libiptc0 ma non √® installabile >>> E: Impossibile correggere i problemi, ci sono pacchetti danneggiati bloccati. >>> test-rtp:/# >>> >>> Praticaly mediaproxy-common depend of libiptc0 but it's not found on debian. >>> >>> The only lib exist are >>> lrwxrwxrwx 1 root root 16 nov 8 2014 /lib/libiptc.so -> libiptc.so.0.0.0 >>> lrwxrwxrwx 1 root root 16 nov 8 2014 /lib/libiptc.so.0 -> libiptc.so.0.0.0 >>> -rw-r--r-- 1 root root 5816 nov 8 2014 /lib/libiptc.so.0.0.0 >>> >>> and i tried to add libiptc0 links too but it dosn't fix.. >>> >>> lrwxrwxrwx 1 root root 16 lug 23 12:49 /lib/libiptc0.so -> libiptc.so.0.0.0 >>> lrwxrwxrwx 1 root root 16 lug 23 12:50 /lib/libiptc0.so.0 -> libiptc.so.0.0.0 >>> >>> Any idea ? >>> >>> Regards >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> -- >> Dan >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dan From bogdan at opensips.org Tue Jul 24 07:58:04 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 24 Jul 2018 14:58:04 +0300 Subject: [OpenSIPS-Users] Users Digest, Vol 120, Issue 29 In-Reply-To: References: Message-ID: Hi Kirill, I just pushed a fix for that extra '\n': https://github.com/OpenSIPS/opensips/commit/01f8f0867431ddb9e085f4cee460e3e748314887 The fix is on all branches, so update from GIT and give it a try. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 07/20/2018 10:29 AM, Kirill Galinurov wrote: > Yes it's wrong Data format. > "Date":"Thu Jul 19 11:30:33 2018 > > 2018-07-19 19:00 GMT+03:00 >: > > Send Users mailing list submissions to > users at lists.opensips.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > or, via email, send a message with subject or body 'help' to > users-request at lists.opensips.org > > > You can reach the person managing the list at > users-owner at lists.opensips.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: mi_json returns invalid JSON (Răzvan Crainea) > 2. Re: mi_json returns invalid JSON (Ben Newlin) > 3. Re: mi_json returns invalid JSON (Ben Newlin) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 19 Jul 2018 10:14:37 +0300 > From: Răzvan Crainea > > To: users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] mi_json returns invalid JSON > Message-ID: <04329915-bc6d-0b21-547f-c114ccca3e66 at opensips.org > > > Content-Type: text/plain; charset=utf-8; format=flowed > > Hi, Kirill! > > This doesn't seem to be an invalid JSON, but only a bad formatting of > the date. I agree it should not include the "\n", because it is > useless, > but this doesn't make it invalid. Are you experiencing trouble with > using JSON libraries that don't support this format? > > Best regards, > Răzvan > > On 07/17/2018 12:58 PM, Kirill Galinurov wrote: > > Hi. I use opensips mi_json module to get statistics. > > But for some commands it returns incorrect JSON. > > For example /json/dr_reload_status returns > > > > {"Partition": [{"value":"sipprovider_def", "attributes":{"Date": > "Mon > > Jul 16 07:10:58 2018 > > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 > 07:11:59 2018 > > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 > 07:11:59 2018 > > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 > 07:12:54 2018 > > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:12:54 2018 > > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 07:13:51 2018 > > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 > 07:13:52 2018 > > "}}]} > > > > This is incorrect, because it shoulde not include "\n" symbols. > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > > > ------------------------------ > > Message: 2 > Date: Thu, 19 Jul 2018 12:58:24 +0000 > From: Ben Newlin > > To: OpenSIPS users mailling list > > Subject: Re: [OpenSIPS-Users] mi_json returns invalid JSON > Message-ID: > > Content-Type: text/plain; charset=UTF-8 > > I believe newline characters used within JSON must be escaped. So > if the JSON actually contains "\n" that is invalid. It must be "\\n". > > Ben Newlin > > On 7/19/18, 3:19 AM, "Users on behalf of Răzvan Crainea" > on behalf of > razvan at opensips.org > wrote: > > Hi, Kirill! > > This doesn't seem to be an invalid JSON, but only a bad > formatting of > the date. I agree it should not include the "\n", because it > is useless, > but this doesn't make it invalid. Are you experiencing trouble > with > using JSON libraries that don't support this format? > > Best regards, > Răzvan > > On 07/17/2018 12:58 PM, Kirill Galinurov wrote: > > Hi. I use opensips mi_json module to get statistics. > > But for some commands it returns incorrect JSON. > > For example /json/dr_reload_status returns > > > > {"Partition": [{"value":"sipprovider_def", > "attributes":{"Date": "Mon > > Jul 16 07:10:58 2018 > > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 > 07:11:59 2018 > > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 > 07:11:59 2018 > > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul 16 > 07:12:54 2018 > > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 > 07:12:54 2018 > > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 > 07:13:51 2018 > > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul 16 > 07:13:52 2018 > > "}}]} > > > > This is incorrect, because it shoulde not include "\n" symbols. > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > ------------------------------ > > Message: 3 > Date: Thu, 19 Jul 2018 13:03:09 +0000 > From: Ben Newlin > > To: OpenSIPS users mailling list > > Subject: Re: [OpenSIPS-Users] mi_json returns invalid JSON > Message-ID: > > Content-Type: text/plain; charset=UTF-8 > > Actually, after further investigation I'm no longer sure that is > true. According to http://www.json.org/ the "\n" is valid in JSON. > > Ben Newlin > > On 7/19/18, 8:58 AM, "Ben Newlin" > wrote: > > I believe newline characters used within JSON must be escaped. > So if the JSON actually contains "\n" that is invalid. It must be > "\\n". > > Ben Newlin > > On 7/19/18, 3:19 AM, "Users on behalf of Răzvan Crainea" > on behalf of > razvan at opensips.org > wrote: > > Hi, Kirill! > > This doesn't seem to be an invalid JSON, but only a bad > formatting of > the date. I agree it should not include the "\n", because > it is useless, > but this doesn't make it invalid. Are you experiencing > trouble with > using JSON libraries that don't support this format? > > Best regards, > Răzvan > > On 07/17/2018 12:58 PM, Kirill Galinurov wrote: > > Hi. I use opensips mi_json module to get statistics. > > But for some commands it returns incorrect JSON. > > For example /json/dr_reload_status returns > > > > {"Partition": [{"value":"sipprovider_def", > "attributes":{"Date": "Mon > > Jul 16 07:10:58 2018 > > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul > 16 07:11:59 2018 > > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul > 16 07:11:59 2018 > > "}}, {"value":"megafon", "attributes":{"Date": "Mon Jul > 16 07:12:54 2018 > > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 > 07:12:54 2018 > > "}}, {"value":"mts", "attributes":{"Date": "Mon Jul 16 > 07:13:51 2018 > > "}}, {"value":"beeline", "attributes":{"Date": "Mon Jul > 16 07:13:52 2018 > > "}}]} > > > > This is incorrect, because it shoulde not include "\n" > symbols. > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ------------------------------ > > End of Users Digest, Vol 120, Issue 29 > ************************************** > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From mimmo at ats.it Tue Jul 24 09:17:28 2018 From: mimmo at ats.it (Simone Sestini) Date: Tue, 24 Jul 2018 15:17:28 +0200 Subject: [OpenSIPS-Users] Mediaproxy and Debian 8.11 In-Reply-To: References: <75F34A6B-EA04-4BEB-B905-8B29E061B50A@ag-projects.com> <3d98da49-f7b0-d3a7-65ae-52ddabf7974c@ats.it> Message-ID: Dear Dan, this is very clear to me.. this way fix the issue .. but i can suggest you and the stuff to update the Doc pages.. --> http://mediaproxy.ag-projects.com/download/ The right way for configure is to add on the source list of apt the follow lines.. where you need to substitude "jessie" with the debian release.. deb http://ftp.debian.org/debian/ jessie-backports main deb-src http://ftp.debian.org/debian/ jessie-backports main deb http://ag-projects.com/debian jessie main deb-src http://ag-projects.com/debian jessie main Thanks a lot for your support. Regards Il 24/07/18 13:27, Dan Pascu ha scritto: > Your problem is that you mix debian 8.11 (jessie) with ag-projects' debian unstable repositories. > > Change your ag-projects repositories to read 'jessie main' instead of 'unstable main', rerun apt-get update and it should work. > > You might also consider adding the jessie-backports repository: > > deb http://ftp.it.debian.org/debian/ jessie-bakcports main > deb-src http://ftp.it.debian.org/debian/ jessie-bakcports main > > Some of our software needs newer versions of some packages that are only available in the jessie-backports repository (not sure if mediaproxy is one of them, but others like blink or sipsimple are). > > On 24 Jul 2018, at 13:48, Simone Sestini wrote: > >> Hi, >> >> only the standard and official repository here.. >> >> ------- >> deb http://ftp.it.debian.org/debian/ jessie main >> deb-src http://ftp.it.debian.org/debian/ jessie main >> >> deb http://security.debian.org/ jessie/updates main >> deb-src http://security.debian.org/ jessie/updates main >> >> # jessie-updates, previously known as 'volatile' >> deb http://ftp.it.debian.org/debian/ jessie-updates main >> deb-src http://ftp.it.debian.org/debian/ jessie-updates main >> >> # AG Projects software >> deb http://ag-projects.com/debian unstable main >> deb-src http://ag-projects.com/debian unstable main >> >> ------- >> >> and here the output for the policy you requested. >> >> test-rtp:~# apt-cache policy iptables mediaproxy-common >> iptables: >> Installato: 1.4.21-2+b1 >> Candidato: 1.4.21-2+b1 >> Tabella versione: >> *** 1.4.21-2+b1 0 >> 500 http://ftp.it.debian.org/debian/ jessie/main amd64 Packages >> 100 /var/lib/dpkg/status >> mediaproxy-common: >> Installato: (nessuno) >> Candidato: 2.6.6 >> Tabella versione: >> 2.6.6 0 >> 500 http://ag-projects.com/debian/ unstable/main amd64 Packages >> >> >> Thanks >> >> >> Il 24/07/18 12:32, Dan Pascu ha scritto: >>> Please paste apt's sources list (you can delete private repositories if you have any, I only want to see debian's and ag-projects' repositories). Alternatively you can paste the output from >>> >>> apt-cache policy iptables mediaproxy-common >>> >>> On 24 Jul 2018, at 12:20, Simone Sestini wrote: >>> >>>> Dear list.. >>>> >>>> I'm trying to install mediaproxy software on Debian 8.11. >>>> >>>> I'm looking around for a solution.. because i'm receiving >>>> test-rtp:/# apt-get install mediaproxy-common mediaproxy-relay mediaproxy-web-sessions >>>> Lettura elenco dei pacchetti... Fatto >>>> Generazione albero delle dipendenze >>>> Lettura informazioni sullo stato... Fatto >>>> Alcuni pacchetti non possono essere installati. Questo pu√≤ voler dire >>>> che √® stata richiesta una situazione impossibile oppure, se si sta >>>> usando una distribuzione in sviluppo, che alcuni pacchetti richiesti >>>> non sono ancora stati creati o sono stati rimossi da Incoming. >>>> Le seguenti informazioni possono aiutare a risolvere la situazione: >>>> >>>> I seguenti pacchetti hanno dipendenze non soddisfatte: >>>> mediaproxy-common : Dipende: libiptc0 ma non √® installabile >>>> E: Impossibile correggere i problemi, ci sono pacchetti danneggiati bloccati. >>>> test-rtp:/# >>>> >>>> Praticaly mediaproxy-common depend of libiptc0 but it's not found on debian. >>>> >>>> The only lib exist are >>>> lrwxrwxrwx 1 root root 16 nov 8 2014 /lib/libiptc.so -> libiptc.so.0.0.0 >>>> lrwxrwxrwx 1 root root 16 nov 8 2014 /lib/libiptc.so.0 -> libiptc.so.0.0.0 >>>> -rw-r--r-- 1 root root 5816 nov 8 2014 /lib/libiptc.so.0.0.0 >>>> >>>> and i tried to add libiptc0 links too but it dosn't fix.. >>>> >>>> lrwxrwxrwx 1 root root 16 lug 23 12:49 /lib/libiptc0.so -> libiptc.so.0.0.0 >>>> lrwxrwxrwx 1 root root 16 lug 23 12:50 /lib/libiptc0.so.0 -> libiptc.so.0.0.0 >>>> >>>> Any idea ? >>>> >>>> Regards >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> -- >>> Dan >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > Dan > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From wallnut.monkeys at gmail.com Tue Jul 24 14:46:19 2018 From: wallnut.monkeys at gmail.com (Dominic) Date: Tue, 24 Jul 2018 14:46:19 -0400 Subject: [OpenSIPS-Users] mid-registrar question Message-ID: Hi all, I'm currently a bit stuck trying to implement a mid-registrar in my dev environment. My setup is that I have a hardphone that is registering to opensips, opensips challenges the register, and once authenticated, the register is sent to an Asterisk server that is setup with no authentication. The diagram below illustrates this: ┌─┐ ║"│ └┬┘ ┌┼┐ │ ┌────────┐ ┌────────┐ ┌┴┐ │OpenSips│ │Asterisk│ Alice └────────┘ └────────┘ │REGISTER (expires 60 sec)│ │ │─────────────────────────> │ │ │ │ │ 401 Unauthorized │ │ │<───────────────────────── │ │ │ │ │REGISTER (expires 60 sec)│ │ │─────────────────────────> │ │ │ │ │ │ REGISTER (expires 3600 sec)│ │ │ ───────────────────────────> │ │ │ │ │ 200 OK (expires 3600 sec) │ │ │ <─────────────────────────── │ │ │ │200 OK (expires 3600 sec)│ │ │<───────────────────────── │ Alice ┌────────┐ ┌────────┐ ┌─┐ │OpenSips│ │Asterisk│ ║"│ └────────┘ └────────┘ └┬┘ ┌┼┐ │ ┌┴┐ I have set the mid_registrar.outgoing_expires to 3600 secs, so Opensips alters the expires in the REGISTER that is sent to asterisk (which is what I want), however, my issue is that the 3600 seconds gets propagated back to Alice, through the 200 OK that is replied from Asterisk to OpenSips which in turn sends it back to Alice. I was wondering if anyone would know of a way to send back 60 seconds in the 200 OK that is sent back to Alice so that Alice keeps sending me REGISTERs every 60 seconds. Thanks Dominic -------------- next part -------------- An HTML attachment was scrubbed... URL: From kurgan-rus at inbox.ru Wed Jul 25 00:41:35 2018 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Wed, 25 Jul 2018 07:41:35 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?mid-registrar_question?= In-Reply-To: References: Message-ID: <1532493695.89917255@f474.i.mail.ru> Hi Dominic! Could you show your script? It seems that there's some misconfiguration. I can share you my own: ... loadmodule "mid_registrar.so" modparam("mid_registrar", "mode", 1) modparam("mid_registrar", "outgoing_expires", 3600) modparam("mid_registrar", "retry_after", 30) ... # REGISTER processing if ( is_method("REGISTER") ) { mid_registrar_save("location"); switch ($retcode) { case 1: $ru = "sip:10.223.15.21:5070"; # this is to be sent to Asterisk route(RELAY); break; case 2: #xlog("L_INFO", "REGISTER has been absorbed!\n"); break; default: xlog("L_ERROR", "failed to save registration! ($$ci=$ci)\n"); } exit; } ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ From kurgan-rus at inbox.ru Wed Jul 25 01:05:29 2018 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Wed, 25 Jul 2018 08:05:29 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?mid-registrar_question?= Message-ID: <1532495129.803446093@f40.i.mail.ru> ... and yes, route[RELAY] is like this: route[RELAY] { if ( !t_relay() ) { send_reply("500","Internal Error"); } exit; } or you may simply do: # REGISTER processing if ( is_method("REGISTER") ) { mid_registrar_save("location"); switch ($retcode) { case 1: $ru = "sip:10.223.15.21:5070"; t_relay(); break; case 2: break; default: xlog("L_ERROR", "failed to save registration! ($$ci=$ci)\n"); } exit; } Here's the tutorial http://www.opensips.org/Documentation/Tutorials-MidRegistrar But keep in mind that it is for OpenSIPS 2.3, as 2.4 does not have 'insertion_mode' option described there. ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ From wallnut.monkeys at gmail.com Wed Jul 25 08:13:24 2018 From: wallnut.monkeys at gmail.com (Dominic) Date: Wed, 25 Jul 2018 08:13:24 -0400 Subject: [OpenSIPS-Users] mid-registrar question In-Reply-To: <1532495129.803446093@f40.i.mail.ru> References: <1532495129.803446093@f40.i.mail.ru> Message-ID: Thanks Alexey, I will give it another shot first thing this morning. On Wed, Jul 25, 2018 at 1:05 AM, Alexey Kazantsev via Users < users at lists.opensips.org> wrote: > ... and yes, route[RELAY] is like this: > > > route[RELAY] { > if ( !t_relay() ) { > send_reply("500","Internal Error"); > } > exit; > } > > > or you may simply do: > > # REGISTER processing > if ( is_method("REGISTER") ) { > mid_registrar_save("location"); > switch ($retcode) { > case 1: > $ru = "sip:10.223.15.21:5070"; > t_relay(); > break; > case 2: > break; > default: > xlog("L_ERROR", "failed to save > registration! ($$ci=$ci)\n"); > } > exit; > } > > > Here's the tutorial http://www.opensips.org/Documentation/Tutorials- > MidRegistrar > But keep in mind that it is for OpenSIPS 2.3, as 2.4 does not have > 'insertion_mode' > option described there. > > > ----------------------------------------------- > BR, Alexey > http://alexeyka.zantsev.com/ > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From wallnut.monkeys at gmail.com Wed Jul 25 09:21:36 2018 From: wallnut.monkeys at gmail.com (Dominic) Date: Wed, 25 Jul 2018 09:21:36 -0400 Subject: [OpenSIPS-Users] mid-registrar question In-Reply-To: References: <1532495129.803446093@f40.i.mail.ru> Message-ID: Alexey I basically started over from scratch using your script as a guide and everything is working fine so far, I wish I could tell what I initially did wrong, but I had a LOT of strange things in there, being a dev box and all. Thanks a LOT for the help On Wed, Jul 25, 2018 at 8:13 AM, Dominic wrote: > Thanks Alexey, I will give it another shot first thing this morning. > > On Wed, Jul 25, 2018 at 1:05 AM, Alexey Kazantsev via Users < > users at lists.opensips.org> wrote: > >> ... and yes, route[RELAY] is like this: >> >> >> route[RELAY] { >> if ( !t_relay() ) { >> send_reply("500","Internal Error"); >> } >> exit; >> } >> >> >> or you may simply do: >> >> # REGISTER processing >> if ( is_method("REGISTER") ) { >> mid_registrar_save("location"); >> switch ($retcode) { >> case 1: >> $ru = "sip:10.223.15.21:5070"; >> t_relay(); >> break; >> case 2: >> break; >> default: >> xlog("L_ERROR", "failed to save >> registration! ($$ci=$ci)\n"); >> } >> exit; >> } >> >> >> Here's the tutorial http://www.opensips.org/Docume >> ntation/Tutorials-MidRegistrar >> But keep in mind that it is for OpenSIPS 2.3, as 2.4 does not have >> 'insertion_mode' >> option described there. >> >> >> ----------------------------------------------- >> BR, Alexey >> http://alexeyka.zantsev.com/ >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Wed Jul 25 11:09:54 2018 From: john.quick at smartvox.co.uk (John Quick) Date: Wed, 25 Jul 2018 16:09:54 +0100 Subject: [OpenSIPS-Users] Using LetsEncrypt certs with v2.4 Message-ID: <002101d42429$8b27afc0$a1770f40$@smartvox.co.uk> Does anyone have experience using LetsEncrypt certificates for tls or wss in OpenSIPS v2.4.x over a long enough period of time for the certificate to be renewed? Does the OpenSIPS service need to be restarted after each certbot renewal? This happens about every 2 months. I have configured opensips so the path in modparam("tls_mgm", "certificate" is "/etc/letsencrypt/live//cert.pem" This is actually a sym-link to the actual cert. It seems to work okay, but I'm wondering what will happen in two months' time when the cert is renewed. Thanks. John Quick Smartvox Limited Web: www.smartvox.co.uk From gmaruzz at gmail.com Wed Jul 25 12:29:47 2018 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 25 Jul 2018 18:29:47 +0200 Subject: [OpenSIPS-Users] FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and Conference Service, ClueCon Presentation Slides Message-ID: Hello fellow RTCers, just finished presenting about scaling videoconferencing, chat and moderation/direction controls in SIP, with web clients, deskphones, smartphone apps, server push (google fcm) via cordova and SIP.js at www.cluecon.com. Here's the slides of my presentation: http://178.79.181.125/Maruzzelli_FreeSWITCH_OpenSIPS_WebRTC_Pure_SIP_Video_Call_Conferencing_and_Chat_ClueCon_2018_Chicago.pdf or, shorter, http://178.79.181.125/Maruzzelli_ClueCon_2018.pdf >From ClueCon blurb: FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and Conference Service On top of traditional SIP deskphones and softphones, WebRTC let us extend the reach of SIP to all browsers, and to smartphone apps that wake up with a server push. OpenSIPS has superb SIP and WebRTC support and can protect, balance and scale FreeSWITCH insuperable conferencing, video MCU, media switching/mixing, SIP SIMPLE message processing capabilities. We will see how to build and implement such a complete platform, touching both on servers and on clients. Hint: SIMPLE, the SIP Instant Messaging Protocol, can be leveraged to transport statuses, commands and chats between participants and administrators of SIP audio/video calls/conferences. Giovanni MaruzzelliPrincipal at OpenTelecom.IT ------------------------------ Giovanni is a consultant for the Telco sector, developing software and training courses for FreeSWITCH, SIP, WebRTC and Kamailio. He is heavily engaged with FreeSWITCH, of which he wrote the interface with Skype and with cellular phones. An Internet tech pioneer, in 1996 Giovanni was cofounder of Italia Online, the most popular Italian portal and consumer ISP, and architect of its Internet technologies – www.italiaonline.it Then supervisor of Internet operations and architect of the first engine for paid access to www.ilsole24ore.com , the most read financial newspaper in Italy and to its databases (migrated from mainframe). After that, he was CEO of venture capital funded Matrice, developing Telemail unified messaging and multi language phone access to email (Text To Speech), and CTO of incubator funded Open4, an open source managed applications provider. Then he was for two years in Serbia as Internet and Telecommunication Investment Expert for World Bank – IFC. Since 2005 he’s based in Italy, and serves ICT and Telco companies worldwide. -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From tijmen at ag-projects.com Thu Jul 26 07:00:37 2018 From: tijmen at ag-projects.com (Tijmen de Mes) Date: Thu, 26 Jul 2018 13:00:37 +0200 Subject: [OpenSIPS-Users] Mediaproxy and Debian 8.11 In-Reply-To: References: <75F34A6B-EA04-4BEB-B905-8B29E061B50A@ag-projects.com> <3d98da49-f7b0-d3a7-65ae-52ddabf7974c@ats.it> Message-ID: Hi Simone, Thanks for your suggestion and we updated the download and install guide. Best regards, Tijmen de Mes — AG Projects > On 24 jul. 2018, at 15:17, Simone Sestini wrote: > > Dear Dan, > > this is very clear to me.. this way fix the issue .. but i can suggest you and the stuff to update the Doc pages.. > > --> http://mediaproxy.ag-projects.com/download/ > > > The right way for configure is to add on the source list of apt the follow lines.. where you need to substitude "jessie" with the debian release.. > > > deb http://ftp.debian.org/debian/ jessie-backports main > deb-src http://ftp.debian.org/debian/ jessie-backports main > > deb http://ag-projects.com/debian jessie main > deb-src http://ag-projects.com/debian jessie main > > > Thanks a lot for your support. > > Regards > > > > Il 24/07/18 13:27, Dan Pascu ha scritto: >> Your problem is that you mix debian 8.11 (jessie) with ag-projects' debian unstable repositories. >> >> Change your ag-projects repositories to read 'jessie main' instead of 'unstable main', rerun apt-get update and it should work. >> >> You might also consider adding the jessie-backports repository: >> >> deb http://ftp.it.debian.org/debian/ jessie-bakcports main >> deb-src http://ftp.it.debian.org/debian/ jessie-bakcports main >> >> Some of our software needs newer versions of some packages that are only available in the jessie-backports repository (not sure if mediaproxy is one of them, but others like blink or sipsimple are). >> >> On 24 Jul 2018, at 13:48, Simone Sestini wrote: >> >>> Hi, >>> >>> only the standard and official repository here.. >>> >>> ------- >>> deb http://ftp.it.debian.org/debian/ jessie main >>> deb-src http://ftp.it.debian.org/debian/ jessie main >>> >>> deb http://security.debian.org/ jessie/updates main >>> deb-src http://security.debian.org/ jessie/updates main >>> >>> # jessie-updates, previously known as 'volatile' >>> deb http://ftp.it.debian.org/debian/ jessie-updates main >>> deb-src http://ftp.it.debian.org/debian/ jessie-updates main >>> >>> # AG Projects software >>> deb http://ag-projects.com/debian unstable main >>> deb-src http://ag-projects.com/debian unstable main >>> >>> ------- >>> >>> and here the output for the policy you requested. >>> >>> test-rtp:~# apt-cache policy iptables mediaproxy-common >>> iptables: >>> Installato: 1.4.21-2+b1 >>> Candidato: 1.4.21-2+b1 >>> Tabella versione: >>> *** 1.4.21-2+b1 0 >>> 500 http://ftp.it.debian.org/debian/ jessie/main amd64 Packages >>> 100 /var/lib/dpkg/status >>> mediaproxy-common: >>> Installato: (nessuno) >>> Candidato: 2.6.6 >>> Tabella versione: >>> 2.6.6 0 >>> 500 http://ag-projects.com/debian/ unstable/main amd64 Packages >>> >>> >>> Thanks >>> >>> >>> Il 24/07/18 12:32, Dan Pascu ha scritto: >>>> Please paste apt's sources list (you can delete private repositories if you have any, I only want to see debian's and ag-projects' repositories). Alternatively you can paste the output from >>>> >>>> apt-cache policy iptables mediaproxy-common >>>> >>>> On 24 Jul 2018, at 12:20, Simone Sestini wrote: >>>> >>>>> Dear list.. >>>>> >>>>> I'm trying to install mediaproxy software on Debian 8.11. >>>>> >>>>> I'm looking around for a solution.. because i'm receiving >>>>> test-rtp:/# apt-get install mediaproxy-common mediaproxy-relay mediaproxy-web-sessions >>>>> Lettura elenco dei pacchetti... Fatto >>>>> Generazione albero delle dipendenze >>>>> Lettura informazioni sullo stato... Fatto >>>>> Alcuni pacchetti non possono essere installati. Questo pu√≤ voler dire >>>>> che √® stata richiesta una situazione impossibile oppure, se si sta >>>>> usando una distribuzione in sviluppo, che alcuni pacchetti richiesti >>>>> non sono ancora stati creati o sono stati rimossi da Incoming. >>>>> Le seguenti informazioni possono aiutare a risolvere la situazione: >>>>> >>>>> I seguenti pacchetti hanno dipendenze non soddisfatte: >>>>> mediaproxy-common : Dipende: libiptc0 ma non √® installabile >>>>> E: Impossibile correggere i problemi, ci sono pacchetti danneggiati bloccati. >>>>> test-rtp:/# >>>>> >>>>> Praticaly mediaproxy-common depend of libiptc0 but it's not found on debian. >>>>> >>>>> The only lib exist are >>>>> lrwxrwxrwx 1 root root 16 nov 8 2014 /lib/libiptc.so -> libiptc.so.0.0.0 >>>>> lrwxrwxrwx 1 root root 16 nov 8 2014 /lib/libiptc.so.0 -> libiptc.so.0.0.0 >>>>> -rw-r--r-- 1 root root 5816 nov 8 2014 /lib/libiptc.so.0.0.0 >>>>> >>>>> and i tried to add libiptc0 links too but it dosn't fix.. >>>>> >>>>> lrwxrwxrwx 1 root root 16 lug 23 12:49 /lib/libiptc0.so -> libiptc.so.0.0.0 >>>>> lrwxrwxrwx 1 root root 16 lug 23 12:50 /lib/libiptc0.so.0 -> libiptc.so.0.0.0 >>>>> >>>>> Any idea ? >>>>> >>>>> Regards >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> -- >>>> Dan >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> -- >> Dan >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Thu Jul 26 07:56:18 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 26 Jul 2018 14:56:18 +0300 Subject: [OpenSIPS-Users] Using LetsEncrypt certs with v2.4 In-Reply-To: <002101d42429$8b27afc0$a1770f40$@smartvox.co.uk> References: <002101d42429$8b27afc0$a1770f40$@smartvox.co.uk> Message-ID: Hi John, When the cert is configured via modparam, the cert is loaded on startup by OpenSIPS, so any renewal of the cert will have 0 impact on OpenSIPS - so you will have to restart after each renewal. I suggest you to provision the certs via DB (and not script), so you can do a reload after renewal, with any need to restart opensips. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 07/25/2018 06:09 PM, John Quick wrote: > Does anyone have experience using LetsEncrypt certificates for tls or wss in > OpenSIPS v2.4.x over a long enough period of time for the certificate to be > renewed? > > Does the OpenSIPS service need to be restarted after each certbot renewal? > This happens about every 2 months. > I have configured opensips so the path in modparam("tls_mgm", "certificate" > is "/etc/letsencrypt/live//cert.pem" > This is actually a sym-link to the actual cert. It seems to work okay, but > I'm wondering what will happen in two months' time when the cert is renewed. > > Thanks. > > John Quick > Smartvox Limited > Web: www.smartvox.co.uk > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Thu Jul 26 07:57:17 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 26 Jul 2018 14:57:17 +0300 Subject: [OpenSIPS-Users] FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and Conference Service, ClueCon Presentation Slides In-Reply-To: References: Message-ID: Hey, that's a good presentation Giovanni, thanks for sharing it with us !! Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 07/25/2018 07:29 PM, Giovanni Maruzzelli wrote: > Hello fellow RTCers, > > just finished presenting about scaling videoconferencing, chat and > moderation/direction controls in SIP, with web clients, deskphones, > smartphone apps, server push (google fcm) via cordova and SIP.js at > www.cluecon.com . > > Here's the slides of my presentation: > > http://178.79.181.125/Maruzzelli_FreeSWITCH_OpenSIPS_WebRTC_Pure_SIP_Video_Call_Conferencing_and_Chat_ClueCon_2018_Chicago.pdf > > or, shorter, > > http://178.79.181.125/Maruzzelli_ClueCon_2018.pdf > > > From ClueCon blurb: > > > FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and > Conference Service > > On top of traditional SIP deskphones and softphones, WebRTC let us > extend the reach of SIP to all browsers, and to smartphone apps that > wake up with a server push. OpenSIPS has superb SIP and WebRTC support > and can protect, balance and scale FreeSWITCH insuperable > conferencing, video MCU, media switching/mixing, SIP SIMPLE message > processing capabilities. We will see how to build and implement such a > complete platform, touching both on servers and on clients. Hint: > SIMPLE, the SIP Instant Messaging Protocol, can be leveraged to > transport statuses, commands and chats between participants and > administrators of SIP audio/video calls/conferences. > > > Giovanni Maruzzelli > > > Principal at OpenTelecom.IT > > ------------------------------------------------------------------------ > > Giovanni is a consultant for the Telco sector, developing software and > training courses for FreeSWITCH, SIP, WebRTC and Kamailio. He is > heavily engaged with FreeSWITCH, of which he wrote the interface with > Skype and with cellular phones. An Internet tech pioneer, in 1996 > Giovanni was cofounder of Italia Online, the most popular Italian > portal and consumer ISP, and architect of its Internet technologies – > www.italiaonline.it Then supervisor of > Internet operations and architect of the first engine for paid access > to www.ilsole24ore.com , the most read > financial newspaper in Italy and to its databases (migrated from > mainframe). After that, he was CEO of venture capital funded Matrice, > developing Telemail unified messaging and multi language phone access > to email (Text To Speech), and CTO of incubator funded Open4, an open > source managed applications provider. Then he was for two years in > Serbia as Internet and Telecommunication Investment Expert for World > Bank – IFC. Since 2005 he’s based in Italy, and serves ICT and Telco > companies worldwide. > > > -giovanni > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Thu Jul 26 13:53:13 2018 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Thu, 26 Jul 2018 14:53:13 -0300 Subject: [OpenSIPS-Users] opensips crash Message-ID: <1532627593.2081.1@smtp.networklab.ca> Hello Everyone, Opensips crashing randomly how possible find what the issue Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6297]: CRITICAL:core:handle_tcp_worker: dead tcp worker 4 (EOF received), pid 6292 Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: INFO:core:handle_sigs: child process 6292 exited by a signal 11 Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: INFO:core:handle_sigs: core was generated Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: INFO:core:handle_sigs: terminating due to SIGCHLD Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6265]: INFO:core:sig_usr: signal 15 received Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6266]: INFO:core:sig_usr: signal 15 received Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6267]: INFO:core:sig_usr: signal 15 received Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: INFO:core:shutdown_opensips: process 15(6279) [SIP receiver udp:private_ip_interface:5060 ] terminated, still waiting for 31 more Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: INFO:core:shutdown_opensips: process 18(6282) [SIP receiver udp:public_ip_interface:5082 ] terminated, still waiting for 30 more Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: INFO:core:shutdown_opensips: process 20(6284) [SIP receiver udp:public_ip_interface:5060 ] terminated, still waiting for 29 more Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: INFO:core:shutdown_opensips: process 22(6286) [SIP receiver udp:public_ip_interface:5060 ] terminated, still waiting for 28 more volga629 From bogdan at opensips.org Fri Jul 27 04:43:51 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 27 Jul 2018 11:43:51 +0300 Subject: [OpenSIPS-Users] opensips crash In-Reply-To: <1532627593.2081.1@smtp.networklab.ca> References: <1532627593.2081.1@smtp.networklab.ca> Message-ID: <082efcd8-6754-3743-d649-398f80dd18b0@opensips.org> Hi, See http://www.opensips.org/Documentation/TroubleShooting-Crash for how to extract the backtrace from the corefile. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 07/26/2018 08:53 PM, volga629 at networklab.ca wrote: > Hello Everyone, > Opensips crashing randomly how possible find what the issue > > > Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6297]: > CRITICAL:core:handle_tcp_worker: dead tcp worker 4 (EOF received), pid > 6292 > Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: INFO:core:handle_sigs: > child process 6292 exited by a signal 11 > Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: INFO:core:handle_sigs: > core was generated > Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: INFO:core:handle_sigs: > terminating due to SIGCHLD > Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6265]: INFO:core:sig_usr: > signal 15 received > Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6266]: INFO:core:sig_usr: > signal 15 received > Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6267]: INFO:core:sig_usr: > signal 15 received > Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: > INFO:core:shutdown_opensips: process 15(6279) [SIP receiver > udp:private_ip_interface:5060 ] terminated, still waiting for 31 more > Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: > INFO:core:shutdown_opensips: process 18(6282) [SIP receiver > udp:public_ip_interface:5082 ] terminated, still waiting for 30 more > Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: > INFO:core:shutdown_opensips: process 20(6284) [SIP receiver > udp:public_ip_interface:5060 ] terminated, still waiting for 29 more > Jul 26 12:11:43 sbc01 /usr/sbin/opensips[6264]: > INFO:core:shutdown_opensips: process 22(6286) [SIP receiver > udp:public_ip_interface:5060 ] terminated, still waiting for 28 more > > volga629 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From darham at hotmail.com Mon Jul 30 04:15:43 2018 From: darham at hotmail.com (Jorge Luis Ortea) Date: Mon, 30 Jul 2018 08:15:43 +0000 Subject: [OpenSIPS-Users] From a moment OpenSIPS ignores one end of the communication Message-ID: Hi all, Very strange case, OpenSIPS in the middle SIP transaction ignores one end of the communication. The schema is very simple, OpenSIPS exchanges SIP traffic between another OpenSIPS (with a passthrough T38 Asterisk behind) and a SBC, during a T38 fax call. The sniffer on the Opensips interfaces shows that the packets are arriving correctly. In the capture can see the packets with red arrow are ignored, not even writes in log, as if this packet did not exist, it always ignores the same SIP packets, all calls are same. Without firewall, selinux, ip or port changes .... A curious detail, from a moment Opensips ignores packets from 10.201.198.60, however after BYE packet seems that it manages SIP packets again. Attachment wireshark capture: http://pasteall.org/pic/show.php?id=2c8da47592429e09e462dae45f41e74c Very Thanks. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: From donat.zenichev at gmail.com Fri Jul 20 02:02:08 2018 From: donat.zenichev at gmail.com (Donat Zenichev) Date: Fri, 20 Jul 2018 09:02:08 +0300 Subject: [OpenSIPS-Users] WARNING:core:utimer_ticker: utimer task already schedualed for In-Reply-To: References: Message-ID: Hi again. Guys I need some hints from you, at least personally, so please take a look at my first message if possible. Thanks in advance and have a nice day. 2018-07-16 10:34 GMT+03:00 Donat Zenichev : > Hi community, recently I've got the problems with sudden termination of > the daemon. > > This bunch of similar rows seems to be some timer job that takes longer > than it was supposed to be: > Jul 14 02:13:10 some-machine opensips[25060]: message repeated 3 times: [ > WARNING:core:timer_ticker: timer task already schedualed for > 10040133060 ms (now 10040134050 ms), it may overlap..] > Jul 14 08:45:29 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063404100 ms (now > 10063404200 ms), it may overlap.. > Jul 14 08:45:29 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063404200 ms (now > 10063404300 ms), it may overlap.. > Jul 14 08:45:29 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063404300 ms (now > 10063404400 ms), it may overlap.. > Jul 14 08:45:30 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063404400 ms (now > 10063404500 ms), it may overlap.. > Jul 14 08:45:30 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063404500 ms (now > 10063404600 ms), it may overlap.. > Jul 14 08:45:30 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063404600 ms (now > 10063404700 ms), it may overlap.. > Jul 14 08:45:30 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063404700 ms (now > 10063404800 ms), it may overlap.. > Jul 14 08:45:30 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063404800 ms (now > 10063404900 ms), it may overlap.. > Jul 14 08:45:30 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063404900 ms (now > 10063405010 ms), it may overlap.. > Jul 14 08:45:30 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063405010 ms (now > 10063405110 ms), it may overlap.. > Jul 14 08:45:30 some-machine opensips[25077]: CRITICAL:core:receive_fd: > EOF on 41 > Jul 14 08:45:30 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063405110 ms (now > 10063405210 ms), it may overlap.. > Jul 14 08:45:30 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063405210 ms (now > 10063405310 ms), it may overlap.. > Jul 14 08:45:30 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063405310 ms (now > 10063405410 ms), it may overlap.. > Jul 14 08:45:31 some-machine opensips[25060]: WARNING:core:utimer_ticker: > utimer task already schedualed for 10063405410 ms (now > 10063405510 ms), it may overlap.. > > Then we have the following row, that means that child process 25075 was > killed by SIGSEGV signal (probably this process tried to appeal to > non-existent memory cell?). > Jul 14 08:45:31 some-machine opensips[25054]: INFO:core:handle_sigs: child > process 25075 exited by a signal 11 > > I can see in the logs lots of rows about the process 25075 that was > treating the nathelper module: > Jul 14 08:45:28 some-machine opensips[25075]: ERROR:nathelper:msg_send: > send() for proto 2 failed > Jul 14 08:45:28 some-machine opensips[25075]: ERROR:nathelper:nh_timer: > sip msg_send failed > Jul 14 08:45:28 some-machine opensips[25075]: ERROR:nathelper:msg_send: > send() for proto 2 failed > Jul 14 08:45:28 some-machine opensips[25075]: ERROR:nathelper:nh_timer: > sip msg_send failed > > Then we have a core dump generated > Jul 14 08:45:31 some-machine opensips[25054]: INFO:core:handle_sigs: core > was generated > > Then we have a SIGCHLD > Jul 14 08:45:31 some-machine opensips[25054]: INFO:core:handle_sigs: > terminating due to SIGCHLD > > I'm not sure how this is considered by opensips daemon (afaik a root > process can ignore such requests) but seems that it caused a termination of > the root process as well. > Because just after that we see, that a SIGTERM (killing) for all the > forked processes had started: > Jul 14 08:45:31 some-machine opensips[25054]: INFO:core:handle_sigs: > terminating due to SIGCHLD > Jul 14 08:45:31 some-machine opensips[25074]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25077]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25070]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25064]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25072]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25062]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25060]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25056]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25067]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25065]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25061]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25063]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25066]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25068]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25076]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25069]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25071]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25073]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25058]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25059]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25057]: INFO:core:sig_usr: signal 15 > received > Jul 14 08:45:31 some-machine opensips[25054]: INFO:core:cleanup: cleanup > > And it's bizarre, since just one killed sub-process triggered such general > termination. > > Since we see that, the process 25075 was related to nathelper module, the > problem perhaps is around that. > I've read several bug articles on the similar issues, and one of the > answers (from Razvan Crainea) seems to be relevant to our story: > > "OpenSIPS pings all users in a single timer run. If there are a lot of > users, this might take a lot, especially if users are using TCP, and > some of them have a stale connection. > Increasing the natping_partitions partitions will basically split all > the users in smaller sets, and will spread them among multiple job runs, > thus multiple processes. Therefore increasing this parameter will > definitely help you. > The range is not a magic value . It depends on the amount of users, > bandwidth, etc. It makes sense to set this parameter from 1 to the > maximum number of processes provisioned - anything else will probably > make timer jobs overlap. " > > For now we do have this parameter set to: > modparam("nathelper", "natping_partitions", 4) > > So it performs nat pinging separately, each part has 1/4 part of users. > > Actually it seems pretty enough to work properly, but seems that nathelper > accumulates the "dead" users (that are not available anymore) and tries to > ping them. But it's just my assumption, because we have to many rows with > the same error (not only this time and with this child process, they are > spread all a log), I mean like that: > Jul 14 08:45:28 some-machine opensips[25075]: ERROR:nathelper:msg_send: > send() for proto 2 failed > Jul 14 08:45:28 some-machine opensips[25075]: ERROR:nathelper:nh_timer: > sip msg_send failed > > And parameters that can have an impact on this situation: > modparam("nathelper", "natping_interval", 30) > modparam("nathelper", "natping_partitions", 4) > > My idea is to increase the partitions, probably to value "8" ? > And probably decrease the interval to 10 seconds. > > What would you advice? > > -- > -- > BR, Donat Zenichev > > -- -- BR, Donat Zenichev -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Jul 17 09:07:08 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 17 Jul 2018 13:07:08 +0000 Subject: [OpenSIPS-Users] Multiple branches for a single AOR In-Reply-To: <64fcb3d3-611e-db28-328b-8aee91def2c9@opensips.org> References: <264A3805-19AF-4DF3-BB74-BFA0E490FC19@genesys.com> <64fcb3d3-611e-db28-328b-8aee91def2c9@opensips.org> Message-ID: <9AADEC51-29B7-4E45-8D82-C67AA0482035@genesys.com> Liviu, Yes, I meant it in the more general sense of re-sending the request, but of course one must be mindful of wording when these terms have specific meanings as this one does! It would indeed be a new transaction. Thanks for catching that! Ben Newlin From: Users on behalf of Liviu Chircu Reply-To: OpenSIPS users mailling list Date: Tuesday, July 17, 2018 at 9:03 AM To: "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] Multiple branches for a single AOR Hi Ben, Just for the sake of completeness, if we do decide to retry a request within a failure route, we will not do a "retransmission", but actually create a brand new transaction (with a unique ";branch=" param) on the SIP proxy's UAC side (the outgoing side). Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 17.07.2018 15:54, Ben Newlin wrote: Gerwin, Specifically, it is the t_relay() call within your failure_route. t_relay() is used to send requests, not responses. The automatic action of failure_route if no retransmission is attempted is to send the reply back upstream [1]. So you don’t have to do anything if that is the functionality you desire. [1] http://www.opensips.org/Documentation/Script-Routes-2-4#toc3 Ben Newlin From: Users on behalf of Liviu Chircu Reply-To: OpenSIPS users mailling list Date: Tuesday, July 17, 2018 at 8:50 AM To: "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] Multiple branches for a single AOR Hi Gerwin, Inside your failure route, you are always attempting a retry of any failed request. This logic conflicts with your initial statement that "The intent is to ensure that the response code gets sent through to A-party". Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 17.07.2018 14:35, Gerwin van de Steeg wrote: Folks, I'm trying to narrow down a 482 Merged Request problem on calls from one SIP device to another via OpenSIPS 2.4.1. Yealink T41P SIP device (A-party), calls via OpenSIPS, to another AOR owned by a Zoiper5 device (B-party). The intent is to ensure that when the B-party rejects the call with a 486 Busy Here, that the response code gets sent through to A-party. However what I'm seeing is the 486 gets sent to OpenSIPS which ACK's it, but doesn't go anywhere from there, and then something causes a second invite to be sent from OpenSIPS to the B-party which then responds of course with 482 Merged Request. The call as it is progressing through the call flow seems to be starting a second branch to the AOR (only one SIP device registered using UDP per AOR). What would be causing that second call so that I can eliminate it and get to the behaviour I'm expecting. Just using a slightly modified residential default config template with websocket support (the problem was noticed using SIP.JS but exists also in generic SIP device to SIP device calls). Image containing sngrep of call: https://imgur.com/RCZXkO6 Subscribers are in the form of @ With an alias setup for an extension number. ie. alfred.anderson at ... = 552 alice.bell at ... = 553 excerpt from opensips.cfg if ($rU==NULL) { # request with no Username in RURI send_reply("484","Address Incomplete"); exit; } $acc_extra(src_ip) = $si; # source IP of the request $acc_leg(caller) = $fu; $acc_leg(callee) = $ru; # apply DB based aliases if (alias_db_lookup("dbaliases")) { xlog("Alias lookup success [$fu/$tu/$ru/$ci]"); } else { xlog("Alias lookup failure [$fu/$tu/$ru/$ci]"); } # do blind callforward lookup if (avp_db_load("rU", "$avp(callfwd)")) { t_reply("181", "Call Is Being Forwarded"); $ru = $avp(callfwd); xlog("forwarded call to: $avp(callfwd)"); route(relay); exit; } # apply transformations from dialplan table dp_translate("0", "$rU/$rU"); # check if the call needs to be routed to freeswitch route(freeswitch); # here we would set the redirect URI if it had one route(lookup); } route[lookup] { script_trace(1, "$rm from $si, rur=$ru", "me"); xlog("route:lookup"); # do lookup with method filtering if (!lookup("location","m")) { xlog("lookup failure"); t_newtran(); if (!db_does_uri_exist()) { xlog("$cfg_line: URI doesn't exist"); send_reply("420", "Bad Extension"); exit; } t_reply("404", "Not Found"); exit; } # when routing via usrloc, log the missed calls also do_accounting("db","missed"); route(relay); } route[freeswitch] { xlog("route:freeswitch"); if (!is_method("INVITE")) { return; } # if the called number begins with the right dialplan redirect it to freeswitch # here we take everythign prefixed with a *, strip it, and send it to freeswitch if ($rU=~"^\*") { strip(1); $du = "sip:10.23.4.192:50600"; route(relay); } } route[relay] { xlog("route:relay: Relaying: method=$rm"); # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("per_branch_ops"); t_on_reply("handle_nat"); t_on_failure("missed_call"); } else if (is_method("BYE|CANCEL")) { # cancel the rtpengine transcoding rtpengine_delete(); } if (!t_relay()) { send_reply("500","Internal Error"); } exit; } branch_route[per_branch_ops] { script_trace(1, "$rm from $si, rur=$ru", "me"); xlog("[$ci/$T_branch_idx] branch_route:per_branch_ops: new branch at $ru\n"); # WebSocket specific handling with NORMAL SDP negotiation # assumes SDP offer in the INVITE from the UAC, and SDP # answer is in 200 OK from the UAS if (!is_method("INVITE") || !has_body("application/sdp")) return; if (isflagset(SRC_WS) && isbflagset(DST_WS)) $var(rtpengine_flags) = "ICE=force-relay DTLS=passive"; else if (isflagset(SRC_WS) && !isbflagset(DST_WS)) $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; else if (!isflagset(SRC_WS) && isbflagset(DST_WS)) $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; else if (!isflagset(SRC_WS) && !isbflagset(DST_WS)) $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; # only enable transcoding if websocket call for now if (isflagset(SRC_WS) || isbflagset(DST_WS)) { rtpengine_offer("$var(rtpengine_flags)"); } } onreply_route[handle_nat] { script_trace(1, "$rm from $si, rur=$ru", "me"); xlog("[$ci/$T_branch_idx] onreply_route:handle_nat: $ru\n"); # WebSocket specific handling with NORMAL SDP negotiation # assumes SDP offer in the INVITE from the UAC, and SDP # answer is in 200 OK from the UAS if (!has_body("application/sdp")) return; if (isflagset(SRC_WS) && isbflagset(DST_WS)) $var(rtpengine_flags) = "ICE=force-relay DTLS=passive"; else if (isflagset(SRC_WS) && !isbflagset(DST_WS)) $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force"; else if (!isflagset(SRC_WS) && isbflagset(DST_WS)) $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; else if (!isflagset(SRC_WS) && !isbflagset(DST_WS)) $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove"; # only enable transcoding if websocket call for now if (isflagset(SRC_WS) || isbflagset(DST_WS)) { rtpengine_answer("$var(rtpengine_flags)"); } } failure_route[missed_call] { script_trace(1, "$rm from $si, rur=$ru", "me"); xlog("[$ci/$T_branch_idx] failure_route:missed_call: incoming failure response to $rm <- $T_reply_code/$T_ruri"); if (t_was_cancelled()) { xlog("[$ci/$T_branch_idx] was cancelled"); exit; } do_accounting("db", "missed"); if (!t_relay()) { send_reply("500","Internal Error"); } else { xlog("[$ci/$T_branch_idx] Relay success $rm/$T_reply_code"); } } Cheers, Gerwin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From red_dra at plugit.net Mon Jul 23 07:21:49 2018 From: red_dra at plugit.net (Laura) Date: Mon, 23 Jul 2018 13:21:49 +0200 Subject: [OpenSIPS-Users] Mediaproxy and Debian 8.11 Message-ID: Dear list.. i'm trying to install mediaproxy software on DEbian 8.11. I'm looking around for a solution.. because i'm receiving test-rtp:/# apt-get install mediaproxy-common  mediaproxy-relay mediaproxy-web-sessions Lettura elenco dei pacchetti... Fatto Generazione albero delle dipendenze Lettura informazioni sullo stato... Fatto Alcuni pacchetti non possono essere installati. Questo pu√≤ voler dire che √® stata richiesta una situazione impossibile oppure, se si sta usando una distribuzione in sviluppo, che alcuni pacchetti richiesti non sono ancora stati creati o sono stati rimossi da Incoming. Le seguenti informazioni possono aiutare a risolvere la situazione: I seguenti pacchetti hanno dipendenze non soddisfatte: * mediaproxy-common : Dipende: ***libiptc0* ma non √® installabile* E: Impossibile correggere i problemi, ci sono pacchetti danneggiati bloccati. test-rtp:/# Praticaly mediaproxy-common depend of *libiptc0 *but it's not found on debian. The only lib exist are lrwxrwxrwx 1 root root   16 nov  8  2014 /lib/libiptc.so -> libiptc.so.0.0.0 lrwxrwxrwx 1 root root   16 nov  8  2014 /lib/libiptc.so.0 -> libiptc.so.0.0.0 -rw-r--r-- 1 root root 5816 nov  8  2014 /lib/libiptc.so.0.0.0 and i tried to add libiptc0 links too but it dosn't fix.. lrwxrwxrwx 1 root root   16 lug 23 12:49 /lib/libiptc0.so -> libiptc.so.0.0.0 lrwxrwxrwx 1 root root   16 lug 23 12:50 /lib/libiptc0.so.0 -> libiptc.so.0.0.0 Any idea ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex.tatham at dmcplc.co.uk Tue Jul 24 07:56:01 2018 From: alex.tatham at dmcplc.co.uk (Alex Tatham) Date: Tue, 24 Jul 2018 11:56:01 +0000 Subject: [OpenSIPS-Users] AG Project - Media Proxy Message-ID: Hello everyone, We have been using AG Projects Media Proxy for several years. Recently we've been experiencing intermittent problems with calls not being initiated and we've traced this to the media proxy component. The frequency of the issue is around 1 in 2000 calls. The following error is logged by the SIP proxy "ERROR: mediaproxy [mediaproxy.c:1528]: send_command(): did timeout waiting for an answer". The following error corresponds in the dispatcher component of the Media Proxy software "Connection to OpenSIPS lost: Connection was closed cleanly.". The problem is temporary and subsequent calls are successful without any intervention. I can't find any information on the WWW about either of these messages. The mediaproxy components are running the latest debian packages as is the corresponding module. I do have some older servers running a much older version of mediaproxy that don't have this fault but I've got to update because of a bug with the relay component on that build. Does anyone have any suggestions. Thanks, Alex Alex Tatham Technical Director ​ T:01233 220 943 E:alex.tatham at dmcplc.co.uk W:www.dmctechnologies.co.uk DMC Technologies formerly HP Technology Unit 5, Invicta Business Centre | Orbital Park | Ashford | Kent | TN24 0HB DMC Technologies is a trading name of Hobbs Parker 2000 Limited Registered Office: Romney House, Orbital Park, Monument Way, Ashford TN24 0HB Registered in England No.04027755 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image737066.png Type: image/png Size: 13052 bytes Desc: image737066.png URL: